Threads.cpp revision 72e3f39146fce4686bd96f11057c051bea376dfb
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/primitives.h> 40#include <audio_utils/format.h> 41#include <audio_utils/minifloat.h> 42 43// NBAIO implementations 44#include <media/nbaio/AudioStreamInSource.h> 45#include <media/nbaio/AudioStreamOutSink.h> 46#include <media/nbaio/MonoPipe.h> 47#include <media/nbaio/MonoPipeReader.h> 48#include <media/nbaio/Pipe.h> 49#include <media/nbaio/PipeReader.h> 50#include <media/nbaio/SourceAudioBufferProvider.h> 51 52#include <powermanager/PowerManager.h> 53 54#include <common_time/cc_helper.h> 55#include <common_time/local_clock.h> 56 57#include "AudioFlinger.h" 58#include "AudioMixer.h" 59#include "BufferProviders.h" 60#include "FastMixer.h" 61#include "FastCapture.h" 62#include "ServiceUtilities.h" 63#include "SchedulingPolicyService.h" 64 65#ifdef ADD_BATTERY_DATA 66#include <media/IMediaPlayerService.h> 67#include <media/IMediaDeathNotifier.h> 68#endif 69 70#ifdef DEBUG_CPU_USAGE 71#include <cpustats/CentralTendencyStatistics.h> 72#include <cpustats/ThreadCpuUsage.h> 73#endif 74 75// ---------------------------------------------------------------------------- 76 77// Note: the following macro is used for extremely verbose logging message. In 78// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 79// 0; but one side effect of this is to turn all LOGV's as well. Some messages 80// are so verbose that we want to suppress them even when we have ALOG_ASSERT 81// turned on. Do not uncomment the #def below unless you really know what you 82// are doing and want to see all of the extremely verbose messages. 83//#define VERY_VERY_VERBOSE_LOGGING 84#ifdef VERY_VERY_VERBOSE_LOGGING 85#define ALOGVV ALOGV 86#else 87#define ALOGVV(a...) do { } while(0) 88#endif 89 90// TODO: Move these macro/inlines to a header file. 91#define max(a, b) ((a) > (b) ? (a) : (b)) 92template <typename T> 93static inline T min(const T& a, const T& b) 94{ 95 return a < b ? a : b; 96} 97 98#ifndef ARRAY_SIZE 99#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 100#endif 101 102namespace android { 103 104// retry counts for buffer fill timeout 105// 50 * ~20msecs = 1 second 106static const int8_t kMaxTrackRetries = 50; 107static const int8_t kMaxTrackStartupRetries = 50; 108// allow less retry attempts on direct output thread. 109// direct outputs can be a scarce resource in audio hardware and should 110// be released as quickly as possible. 111static const int8_t kMaxTrackRetriesDirect = 2; 112 113// don't warn about blocked writes or record buffer overflows more often than this 114static const nsecs_t kWarningThrottleNs = seconds(5); 115 116// RecordThread loop sleep time upon application overrun or audio HAL read error 117static const int kRecordThreadSleepUs = 5000; 118 119// maximum time to wait in sendConfigEvent_l() for a status to be received 120static const nsecs_t kConfigEventTimeoutNs = seconds(2); 121 122// minimum sleep time for the mixer thread loop when tracks are active but in underrun 123static const uint32_t kMinThreadSleepTimeUs = 5000; 124// maximum divider applied to the active sleep time in the mixer thread loop 125static const uint32_t kMaxThreadSleepTimeShift = 2; 126 127// minimum normal sink buffer size, expressed in milliseconds rather than frames 128static const uint32_t kMinNormalSinkBufferSizeMs = 20; 129// maximum normal sink buffer size 130static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 131 132// Offloaded output thread standby delay: allows track transition without going to standby 133static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 134 135// Whether to use fast mixer 136static const enum { 137 FastMixer_Never, // never initialize or use: for debugging only 138 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 139 // normal mixer multiplier is 1 140 FastMixer_Static, // initialize if needed, then use all the time if initialized, 141 // multiplier is calculated based on min & max normal mixer buffer size 142 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 143 // multiplier is calculated based on min & max normal mixer buffer size 144 // FIXME for FastMixer_Dynamic: 145 // Supporting this option will require fixing HALs that can't handle large writes. 146 // For example, one HAL implementation returns an error from a large write, 147 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 148 // We could either fix the HAL implementations, or provide a wrapper that breaks 149 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 150} kUseFastMixer = FastMixer_Static; 151 152// Whether to use fast capture 153static const enum { 154 FastCapture_Never, // never initialize or use: for debugging only 155 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 156 FastCapture_Static, // initialize if needed, then use all the time if initialized 157} kUseFastCapture = FastCapture_Static; 158 159// Priorities for requestPriority 160static const int kPriorityAudioApp = 2; 161static const int kPriorityFastMixer = 3; 162static const int kPriorityFastCapture = 3; 163 164// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 165// for the track. The client then sub-divides this into smaller buffers for its use. 166// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 167// So for now we just assume that client is double-buffered for fast tracks. 168// FIXME It would be better for client to tell AudioFlinger the value of N, 169// so AudioFlinger could allocate the right amount of memory. 170// See the client's minBufCount and mNotificationFramesAct calculations for details. 171 172// This is the default value, if not specified by property. 173static const int kFastTrackMultiplier = 2; 174 175// The minimum and maximum allowed values 176static const int kFastTrackMultiplierMin = 1; 177static const int kFastTrackMultiplierMax = 2; 178 179// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 180static int sFastTrackMultiplier = kFastTrackMultiplier; 181 182// See Thread::readOnlyHeap(). 183// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 184// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 185// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 186static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 187 188// ---------------------------------------------------------------------------- 189 190static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 191 192static void sFastTrackMultiplierInit() 193{ 194 char value[PROPERTY_VALUE_MAX]; 195 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 196 char *endptr; 197 unsigned long ul = strtoul(value, &endptr, 0); 198 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 199 sFastTrackMultiplier = (int) ul; 200 } 201 } 202} 203 204// ---------------------------------------------------------------------------- 205 206#ifdef ADD_BATTERY_DATA 207// To collect the amplifier usage 208static void addBatteryData(uint32_t params) { 209 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 210 if (service == NULL) { 211 // it already logged 212 return; 213 } 214 215 service->addBatteryData(params); 216} 217#endif 218 219 220// ---------------------------------------------------------------------------- 221// CPU Stats 222// ---------------------------------------------------------------------------- 223 224class CpuStats { 225public: 226 CpuStats(); 227 void sample(const String8 &title); 228#ifdef DEBUG_CPU_USAGE 229private: 230 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 231 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 232 233 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 234 235 int mCpuNum; // thread's current CPU number 236 int mCpukHz; // frequency of thread's current CPU in kHz 237#endif 238}; 239 240CpuStats::CpuStats() 241#ifdef DEBUG_CPU_USAGE 242 : mCpuNum(-1), mCpukHz(-1) 243#endif 244{ 245} 246 247void CpuStats::sample(const String8 &title 248#ifndef DEBUG_CPU_USAGE 249 __unused 250#endif 251 ) { 252#ifdef DEBUG_CPU_USAGE 253 // get current thread's delta CPU time in wall clock ns 254 double wcNs; 255 bool valid = mCpuUsage.sampleAndEnable(wcNs); 256 257 // record sample for wall clock statistics 258 if (valid) { 259 mWcStats.sample(wcNs); 260 } 261 262 // get the current CPU number 263 int cpuNum = sched_getcpu(); 264 265 // get the current CPU frequency in kHz 266 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 267 268 // check if either CPU number or frequency changed 269 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 270 mCpuNum = cpuNum; 271 mCpukHz = cpukHz; 272 // ignore sample for purposes of cycles 273 valid = false; 274 } 275 276 // if no change in CPU number or frequency, then record sample for cycle statistics 277 if (valid && mCpukHz > 0) { 278 double cycles = wcNs * cpukHz * 0.000001; 279 mHzStats.sample(cycles); 280 } 281 282 unsigned n = mWcStats.n(); 283 // mCpuUsage.elapsed() is expensive, so don't call it every loop 284 if ((n & 127) == 1) { 285 long long elapsed = mCpuUsage.elapsed(); 286 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 287 double perLoop = elapsed / (double) n; 288 double perLoop100 = perLoop * 0.01; 289 double perLoop1k = perLoop * 0.001; 290 double mean = mWcStats.mean(); 291 double stddev = mWcStats.stddev(); 292 double minimum = mWcStats.minimum(); 293 double maximum = mWcStats.maximum(); 294 double meanCycles = mHzStats.mean(); 295 double stddevCycles = mHzStats.stddev(); 296 double minCycles = mHzStats.minimum(); 297 double maxCycles = mHzStats.maximum(); 298 mCpuUsage.resetElapsed(); 299 mWcStats.reset(); 300 mHzStats.reset(); 301 ALOGD("CPU usage for %s over past %.1f secs\n" 302 " (%u mixer loops at %.1f mean ms per loop):\n" 303 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 304 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 305 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 306 title.string(), 307 elapsed * .000000001, n, perLoop * .000001, 308 mean * .001, 309 stddev * .001, 310 minimum * .001, 311 maximum * .001, 312 mean / perLoop100, 313 stddev / perLoop100, 314 minimum / perLoop100, 315 maximum / perLoop100, 316 meanCycles / perLoop1k, 317 stddevCycles / perLoop1k, 318 minCycles / perLoop1k, 319 maxCycles / perLoop1k); 320 321 } 322 } 323#endif 324}; 325 326// ---------------------------------------------------------------------------- 327// ThreadBase 328// ---------------------------------------------------------------------------- 329 330// static 331const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 332{ 333 switch (type) { 334 case MIXER: 335 return "MIXER"; 336 case DIRECT: 337 return "DIRECT"; 338 case DUPLICATING: 339 return "DUPLICATING"; 340 case RECORD: 341 return "RECORD"; 342 case OFFLOAD: 343 return "OFFLOAD"; 344 default: 345 return "unknown"; 346 } 347} 348 349String8 devicesToString(audio_devices_t devices) 350{ 351 static const struct mapping { 352 audio_devices_t mDevices; 353 const char * mString; 354 } mappingsOut[] = { 355 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 356 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 357 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 358 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 359 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 360 AUDIO_DEVICE_NONE, "NONE", // must be last 361 }, mappingsIn[] = { 362 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 363 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 364 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 365 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 366 AUDIO_DEVICE_NONE, "NONE", // must be last 367 }; 368 String8 result; 369 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 370 const mapping *entry; 371 if (devices & AUDIO_DEVICE_BIT_IN) { 372 devices &= ~AUDIO_DEVICE_BIT_IN; 373 entry = mappingsIn; 374 } else { 375 entry = mappingsOut; 376 } 377 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 378 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 379 if (devices & entry->mDevices) { 380 if (!result.isEmpty()) { 381 result.append("|"); 382 } 383 result.append(entry->mString); 384 } 385 } 386 if (devices & ~allDevices) { 387 if (!result.isEmpty()) { 388 result.append("|"); 389 } 390 result.appendFormat("0x%X", devices & ~allDevices); 391 } 392 if (result.isEmpty()) { 393 result.append(entry->mString); 394 } 395 return result; 396} 397 398String8 inputFlagsToString(audio_input_flags_t flags) 399{ 400 static const struct mapping { 401 audio_input_flags_t mFlag; 402 const char * mString; 403 } mappings[] = { 404 AUDIO_INPUT_FLAG_FAST, "FAST", 405 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 406 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 407 }; 408 String8 result; 409 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 410 const mapping *entry; 411 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 412 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 413 if (flags & entry->mFlag) { 414 if (!result.isEmpty()) { 415 result.append("|"); 416 } 417 result.append(entry->mString); 418 } 419 } 420 if (flags & ~allFlags) { 421 if (!result.isEmpty()) { 422 result.append("|"); 423 } 424 result.appendFormat("0x%X", flags & ~allFlags); 425 } 426 if (result.isEmpty()) { 427 result.append(entry->mString); 428 } 429 return result; 430} 431 432String8 outputFlagsToString(audio_output_flags_t flags) 433{ 434 static const struct mapping { 435 audio_output_flags_t mFlag; 436 const char * mString; 437 } mappings[] = { 438 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 439 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 440 AUDIO_OUTPUT_FLAG_FAST, "FAST", 441 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 442 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 443 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 444 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 445 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 446 }; 447 String8 result; 448 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 449 const mapping *entry; 450 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 451 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 452 if (flags & entry->mFlag) { 453 if (!result.isEmpty()) { 454 result.append("|"); 455 } 456 result.append(entry->mString); 457 } 458 } 459 if (flags & ~allFlags) { 460 if (!result.isEmpty()) { 461 result.append("|"); 462 } 463 result.appendFormat("0x%X", flags & ~allFlags); 464 } 465 if (result.isEmpty()) { 466 result.append(entry->mString); 467 } 468 return result; 469} 470 471const char *sourceToString(audio_source_t source) 472{ 473 switch (source) { 474 case AUDIO_SOURCE_DEFAULT: return "default"; 475 case AUDIO_SOURCE_MIC: return "mic"; 476 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 477 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 478 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 479 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 480 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 481 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 482 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 483 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 484 case AUDIO_SOURCE_HOTWORD: return "hotword"; 485 default: return "unknown"; 486 } 487} 488 489AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 490 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 491 : Thread(false /*canCallJava*/), 492 mType(type), 493 mAudioFlinger(audioFlinger), 494 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 495 // are set by PlaybackThread::readOutputParameters_l() or 496 // RecordThread::readInputParameters_l() 497 //FIXME: mStandby should be true here. Is this some kind of hack? 498 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 499 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 500 // mName will be set by concrete (non-virtual) subclass 501 mDeathRecipient(new PMDeathRecipient(this)), 502 mSystemReady(systemReady) 503{ 504 memset(&mPatch, 0, sizeof(struct audio_patch)); 505} 506 507AudioFlinger::ThreadBase::~ThreadBase() 508{ 509 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 510 mConfigEvents.clear(); 511 512 // do not lock the mutex in destructor 513 releaseWakeLock_l(); 514 if (mPowerManager != 0) { 515 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 516 binder->unlinkToDeath(mDeathRecipient); 517 } 518} 519 520status_t AudioFlinger::ThreadBase::readyToRun() 521{ 522 status_t status = initCheck(); 523 if (status == NO_ERROR) { 524 ALOGI("AudioFlinger's thread %p ready to run", this); 525 } else { 526 ALOGE("No working audio driver found."); 527 } 528 return status; 529} 530 531void AudioFlinger::ThreadBase::exit() 532{ 533 ALOGV("ThreadBase::exit"); 534 // do any cleanup required for exit to succeed 535 preExit(); 536 { 537 // This lock prevents the following race in thread (uniprocessor for illustration): 538 // if (!exitPending()) { 539 // // context switch from here to exit() 540 // // exit() calls requestExit(), what exitPending() observes 541 // // exit() calls signal(), which is dropped since no waiters 542 // // context switch back from exit() to here 543 // mWaitWorkCV.wait(...); 544 // // now thread is hung 545 // } 546 AutoMutex lock(mLock); 547 requestExit(); 548 mWaitWorkCV.broadcast(); 549 } 550 // When Thread::requestExitAndWait is made virtual and this method is renamed to 551 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 552 requestExitAndWait(); 553} 554 555status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 556{ 557 status_t status; 558 559 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 560 Mutex::Autolock _l(mLock); 561 562 return sendSetParameterConfigEvent_l(keyValuePairs); 563} 564 565// sendConfigEvent_l() must be called with ThreadBase::mLock held 566// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 567status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 568{ 569 status_t status = NO_ERROR; 570 571 if (event->mRequiresSystemReady && !mSystemReady) { 572 event->mWaitStatus = false; 573 mPendingConfigEvents.add(event); 574 return status; 575 } 576 mConfigEvents.add(event); 577 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 578 mWaitWorkCV.signal(); 579 mLock.unlock(); 580 { 581 Mutex::Autolock _l(event->mLock); 582 while (event->mWaitStatus) { 583 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 584 event->mStatus = TIMED_OUT; 585 event->mWaitStatus = false; 586 } 587 } 588 status = event->mStatus; 589 } 590 mLock.lock(); 591 return status; 592} 593 594void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event) 595{ 596 Mutex::Autolock _l(mLock); 597 sendIoConfigEvent_l(event); 598} 599 600// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 601void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event) 602{ 603 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event); 604 sendConfigEvent_l(configEvent); 605} 606 607void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 608{ 609 Mutex::Autolock _l(mLock); 610 sendPrioConfigEvent_l(pid, tid, prio); 611} 612 613// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 614void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 615{ 616 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 617 sendConfigEvent_l(configEvent); 618} 619 620// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 621status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 622{ 623 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 624 return sendConfigEvent_l(configEvent); 625} 626 627status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 628 const struct audio_patch *patch, 629 audio_patch_handle_t *handle) 630{ 631 Mutex::Autolock _l(mLock); 632 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 633 status_t status = sendConfigEvent_l(configEvent); 634 if (status == NO_ERROR) { 635 CreateAudioPatchConfigEventData *data = 636 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 637 *handle = data->mHandle; 638 } 639 return status; 640} 641 642status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 643 const audio_patch_handle_t handle) 644{ 645 Mutex::Autolock _l(mLock); 646 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 647 return sendConfigEvent_l(configEvent); 648} 649 650 651// post condition: mConfigEvents.isEmpty() 652void AudioFlinger::ThreadBase::processConfigEvents_l() 653{ 654 bool configChanged = false; 655 656 while (!mConfigEvents.isEmpty()) { 657 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 658 sp<ConfigEvent> event = mConfigEvents[0]; 659 mConfigEvents.removeAt(0); 660 switch (event->mType) { 661 case CFG_EVENT_PRIO: { 662 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 663 // FIXME Need to understand why this has to be done asynchronously 664 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 665 true /*asynchronous*/); 666 if (err != 0) { 667 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 668 data->mPrio, data->mPid, data->mTid, err); 669 } 670 } break; 671 case CFG_EVENT_IO: { 672 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 673 ioConfigChanged(data->mEvent); 674 } break; 675 case CFG_EVENT_SET_PARAMETER: { 676 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 677 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 678 configChanged = true; 679 } 680 } break; 681 case CFG_EVENT_CREATE_AUDIO_PATCH: { 682 CreateAudioPatchConfigEventData *data = 683 (CreateAudioPatchConfigEventData *)event->mData.get(); 684 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 685 } break; 686 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 687 ReleaseAudioPatchConfigEventData *data = 688 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 689 event->mStatus = releaseAudioPatch_l(data->mHandle); 690 } break; 691 default: 692 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 693 break; 694 } 695 { 696 Mutex::Autolock _l(event->mLock); 697 if (event->mWaitStatus) { 698 event->mWaitStatus = false; 699 event->mCond.signal(); 700 } 701 } 702 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 703 } 704 705 if (configChanged) { 706 cacheParameters_l(); 707 } 708} 709 710String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 711 String8 s; 712 if (output) { 713 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 714 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 715 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 716 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 717 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 718 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 719 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 720 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 721 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 722 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 723 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 724 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 725 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 726 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 727 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 728 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 729 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 730 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 731 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 732 } else { 733 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 734 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 735 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 736 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 737 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 738 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 739 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 740 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 741 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 742 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 743 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 744 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 745 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 746 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 747 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 748 } 749 int len = s.length(); 750 if (s.length() > 2) { 751 char *str = s.lockBuffer(len); 752 s.unlockBuffer(len - 2); 753 } 754 return s; 755} 756 757void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 758{ 759 const size_t SIZE = 256; 760 char buffer[SIZE]; 761 String8 result; 762 763 bool locked = AudioFlinger::dumpTryLock(mLock); 764 if (!locked) { 765 dprintf(fd, "thread %p may be deadlocked\n", this); 766 } 767 768 dprintf(fd, " Thread name: %s\n", mThreadName); 769 dprintf(fd, " I/O handle: %d\n", mId); 770 dprintf(fd, " TID: %d\n", getTid()); 771 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 772 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 773 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 774 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 775 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 776 dprintf(fd, " Channel count: %u\n", mChannelCount); 777 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 778 channelMaskToString(mChannelMask, mType != RECORD).string()); 779 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 780 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 781 dprintf(fd, " Pending config events:"); 782 size_t numConfig = mConfigEvents.size(); 783 if (numConfig) { 784 for (size_t i = 0; i < numConfig; i++) { 785 mConfigEvents[i]->dump(buffer, SIZE); 786 dprintf(fd, "\n %s", buffer); 787 } 788 dprintf(fd, "\n"); 789 } else { 790 dprintf(fd, " none\n"); 791 } 792 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 793 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 794 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 795 796 if (locked) { 797 mLock.unlock(); 798 } 799} 800 801void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 802{ 803 const size_t SIZE = 256; 804 char buffer[SIZE]; 805 String8 result; 806 807 size_t numEffectChains = mEffectChains.size(); 808 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 809 write(fd, buffer, strlen(buffer)); 810 811 for (size_t i = 0; i < numEffectChains; ++i) { 812 sp<EffectChain> chain = mEffectChains[i]; 813 if (chain != 0) { 814 chain->dump(fd, args); 815 } 816 } 817} 818 819void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 820{ 821 Mutex::Autolock _l(mLock); 822 acquireWakeLock_l(uid); 823} 824 825String16 AudioFlinger::ThreadBase::getWakeLockTag() 826{ 827 switch (mType) { 828 case MIXER: 829 return String16("AudioMix"); 830 case DIRECT: 831 return String16("AudioDirectOut"); 832 case DUPLICATING: 833 return String16("AudioDup"); 834 case RECORD: 835 return String16("AudioIn"); 836 case OFFLOAD: 837 return String16("AudioOffload"); 838 default: 839 ALOG_ASSERT(false); 840 return String16("AudioUnknown"); 841 } 842} 843 844void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 845{ 846 getPowerManager_l(); 847 if (mPowerManager != 0) { 848 sp<IBinder> binder = new BBinder(); 849 status_t status; 850 if (uid >= 0) { 851 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 852 binder, 853 getWakeLockTag(), 854 String16("media"), 855 uid, 856 true /* FIXME force oneway contrary to .aidl */); 857 } else { 858 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 859 binder, 860 getWakeLockTag(), 861 String16("media"), 862 true /* FIXME force oneway contrary to .aidl */); 863 } 864 if (status == NO_ERROR) { 865 mWakeLockToken = binder; 866 } 867 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 868 } 869} 870 871void AudioFlinger::ThreadBase::releaseWakeLock() 872{ 873 Mutex::Autolock _l(mLock); 874 releaseWakeLock_l(); 875} 876 877void AudioFlinger::ThreadBase::releaseWakeLock_l() 878{ 879 if (mWakeLockToken != 0) { 880 ALOGV("releaseWakeLock_l() %s", mThreadName); 881 if (mPowerManager != 0) { 882 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 883 true /* FIXME force oneway contrary to .aidl */); 884 } 885 mWakeLockToken.clear(); 886 } 887} 888 889void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 890 Mutex::Autolock _l(mLock); 891 updateWakeLockUids_l(uids); 892} 893 894void AudioFlinger::ThreadBase::getPowerManager_l() { 895 if (mSystemReady && mPowerManager == 0) { 896 // use checkService() to avoid blocking if power service is not up yet 897 sp<IBinder> binder = 898 defaultServiceManager()->checkService(String16("power")); 899 if (binder == 0) { 900 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 901 } else { 902 mPowerManager = interface_cast<IPowerManager>(binder); 903 binder->linkToDeath(mDeathRecipient); 904 } 905 } 906} 907 908void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 909 getPowerManager_l(); 910 if (mWakeLockToken == NULL) { 911 ALOGE("no wake lock to update!"); 912 return; 913 } 914 if (mPowerManager != 0) { 915 sp<IBinder> binder = new BBinder(); 916 status_t status; 917 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 918 true /* FIXME force oneway contrary to .aidl */); 919 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 920 } 921} 922 923void AudioFlinger::ThreadBase::clearPowerManager() 924{ 925 Mutex::Autolock _l(mLock); 926 releaseWakeLock_l(); 927 mPowerManager.clear(); 928} 929 930void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 931{ 932 sp<ThreadBase> thread = mThread.promote(); 933 if (thread != 0) { 934 thread->clearPowerManager(); 935 } 936 ALOGW("power manager service died !!!"); 937} 938 939void AudioFlinger::ThreadBase::setEffectSuspended( 940 const effect_uuid_t *type, bool suspend, int sessionId) 941{ 942 Mutex::Autolock _l(mLock); 943 setEffectSuspended_l(type, suspend, sessionId); 944} 945 946void AudioFlinger::ThreadBase::setEffectSuspended_l( 947 const effect_uuid_t *type, bool suspend, int sessionId) 948{ 949 sp<EffectChain> chain = getEffectChain_l(sessionId); 950 if (chain != 0) { 951 if (type != NULL) { 952 chain->setEffectSuspended_l(type, suspend); 953 } else { 954 chain->setEffectSuspendedAll_l(suspend); 955 } 956 } 957 958 updateSuspendedSessions_l(type, suspend, sessionId); 959} 960 961void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 962{ 963 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 964 if (index < 0) { 965 return; 966 } 967 968 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 969 mSuspendedSessions.valueAt(index); 970 971 for (size_t i = 0; i < sessionEffects.size(); i++) { 972 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 973 for (int j = 0; j < desc->mRefCount; j++) { 974 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 975 chain->setEffectSuspendedAll_l(true); 976 } else { 977 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 978 desc->mType.timeLow); 979 chain->setEffectSuspended_l(&desc->mType, true); 980 } 981 } 982 } 983} 984 985void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 986 bool suspend, 987 int sessionId) 988{ 989 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 990 991 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 992 993 if (suspend) { 994 if (index >= 0) { 995 sessionEffects = mSuspendedSessions.valueAt(index); 996 } else { 997 mSuspendedSessions.add(sessionId, sessionEffects); 998 } 999 } else { 1000 if (index < 0) { 1001 return; 1002 } 1003 sessionEffects = mSuspendedSessions.valueAt(index); 1004 } 1005 1006 1007 int key = EffectChain::kKeyForSuspendAll; 1008 if (type != NULL) { 1009 key = type->timeLow; 1010 } 1011 index = sessionEffects.indexOfKey(key); 1012 1013 sp<SuspendedSessionDesc> desc; 1014 if (suspend) { 1015 if (index >= 0) { 1016 desc = sessionEffects.valueAt(index); 1017 } else { 1018 desc = new SuspendedSessionDesc(); 1019 if (type != NULL) { 1020 desc->mType = *type; 1021 } 1022 sessionEffects.add(key, desc); 1023 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1024 } 1025 desc->mRefCount++; 1026 } else { 1027 if (index < 0) { 1028 return; 1029 } 1030 desc = sessionEffects.valueAt(index); 1031 if (--desc->mRefCount == 0) { 1032 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1033 sessionEffects.removeItemsAt(index); 1034 if (sessionEffects.isEmpty()) { 1035 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1036 sessionId); 1037 mSuspendedSessions.removeItem(sessionId); 1038 } 1039 } 1040 } 1041 if (!sessionEffects.isEmpty()) { 1042 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1043 } 1044} 1045 1046void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1047 bool enabled, 1048 int sessionId) 1049{ 1050 Mutex::Autolock _l(mLock); 1051 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1052} 1053 1054void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1055 bool enabled, 1056 int sessionId) 1057{ 1058 if (mType != RECORD) { 1059 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1060 // another session. This gives the priority to well behaved effect control panels 1061 // and applications not using global effects. 1062 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1063 // global effects 1064 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1065 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1066 } 1067 } 1068 1069 sp<EffectChain> chain = getEffectChain_l(sessionId); 1070 if (chain != 0) { 1071 chain->checkSuspendOnEffectEnabled(effect, enabled); 1072 } 1073} 1074 1075// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1076sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1077 const sp<AudioFlinger::Client>& client, 1078 const sp<IEffectClient>& effectClient, 1079 int32_t priority, 1080 int sessionId, 1081 effect_descriptor_t *desc, 1082 int *enabled, 1083 status_t *status) 1084{ 1085 sp<EffectModule> effect; 1086 sp<EffectHandle> handle; 1087 status_t lStatus; 1088 sp<EffectChain> chain; 1089 bool chainCreated = false; 1090 bool effectCreated = false; 1091 bool effectRegistered = false; 1092 1093 lStatus = initCheck(); 1094 if (lStatus != NO_ERROR) { 1095 ALOGW("createEffect_l() Audio driver not initialized."); 1096 goto Exit; 1097 } 1098 1099 // Reject any effect on Direct output threads for now, since the format of 1100 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1101 if (mType == DIRECT) { 1102 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1103 desc->name, mThreadName); 1104 lStatus = BAD_VALUE; 1105 goto Exit; 1106 } 1107 1108 // Reject any effect on mixer or duplicating multichannel sinks. 1109 // TODO: fix both format and multichannel issues with effects. 1110 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1111 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1112 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1113 lStatus = BAD_VALUE; 1114 goto Exit; 1115 } 1116 1117 // Allow global effects only on offloaded and mixer threads 1118 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1119 switch (mType) { 1120 case MIXER: 1121 case OFFLOAD: 1122 break; 1123 case DIRECT: 1124 case DUPLICATING: 1125 case RECORD: 1126 default: 1127 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1128 desc->name, mThreadName); 1129 lStatus = BAD_VALUE; 1130 goto Exit; 1131 } 1132 } 1133 1134 // Only Pre processor effects are allowed on input threads and only on input threads 1135 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1136 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1137 desc->name, desc->flags, mType); 1138 lStatus = BAD_VALUE; 1139 goto Exit; 1140 } 1141 1142 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1143 1144 { // scope for mLock 1145 Mutex::Autolock _l(mLock); 1146 1147 // check for existing effect chain with the requested audio session 1148 chain = getEffectChain_l(sessionId); 1149 if (chain == 0) { 1150 // create a new chain for this session 1151 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1152 chain = new EffectChain(this, sessionId); 1153 addEffectChain_l(chain); 1154 chain->setStrategy(getStrategyForSession_l(sessionId)); 1155 chainCreated = true; 1156 } else { 1157 effect = chain->getEffectFromDesc_l(desc); 1158 } 1159 1160 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1161 1162 if (effect == 0) { 1163 int id = mAudioFlinger->nextUniqueId(); 1164 // Check CPU and memory usage 1165 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1166 if (lStatus != NO_ERROR) { 1167 goto Exit; 1168 } 1169 effectRegistered = true; 1170 // create a new effect module if none present in the chain 1171 effect = new EffectModule(this, chain, desc, id, sessionId); 1172 lStatus = effect->status(); 1173 if (lStatus != NO_ERROR) { 1174 goto Exit; 1175 } 1176 effect->setOffloaded(mType == OFFLOAD, mId); 1177 1178 lStatus = chain->addEffect_l(effect); 1179 if (lStatus != NO_ERROR) { 1180 goto Exit; 1181 } 1182 effectCreated = true; 1183 1184 effect->setDevice(mOutDevice); 1185 effect->setDevice(mInDevice); 1186 effect->setMode(mAudioFlinger->getMode()); 1187 effect->setAudioSource(mAudioSource); 1188 } 1189 // create effect handle and connect it to effect module 1190 handle = new EffectHandle(effect, client, effectClient, priority); 1191 lStatus = handle->initCheck(); 1192 if (lStatus == OK) { 1193 lStatus = effect->addHandle(handle.get()); 1194 } 1195 if (enabled != NULL) { 1196 *enabled = (int)effect->isEnabled(); 1197 } 1198 } 1199 1200Exit: 1201 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1202 Mutex::Autolock _l(mLock); 1203 if (effectCreated) { 1204 chain->removeEffect_l(effect); 1205 } 1206 if (effectRegistered) { 1207 AudioSystem::unregisterEffect(effect->id()); 1208 } 1209 if (chainCreated) { 1210 removeEffectChain_l(chain); 1211 } 1212 handle.clear(); 1213 } 1214 1215 *status = lStatus; 1216 return handle; 1217} 1218 1219sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1220{ 1221 Mutex::Autolock _l(mLock); 1222 return getEffect_l(sessionId, effectId); 1223} 1224 1225sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1226{ 1227 sp<EffectChain> chain = getEffectChain_l(sessionId); 1228 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1229} 1230 1231// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1232// PlaybackThread::mLock held 1233status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1234{ 1235 // check for existing effect chain with the requested audio session 1236 int sessionId = effect->sessionId(); 1237 sp<EffectChain> chain = getEffectChain_l(sessionId); 1238 bool chainCreated = false; 1239 1240 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1241 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1242 this, effect->desc().name, effect->desc().flags); 1243 1244 if (chain == 0) { 1245 // create a new chain for this session 1246 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1247 chain = new EffectChain(this, sessionId); 1248 addEffectChain_l(chain); 1249 chain->setStrategy(getStrategyForSession_l(sessionId)); 1250 chainCreated = true; 1251 } 1252 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1253 1254 if (chain->getEffectFromId_l(effect->id()) != 0) { 1255 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1256 this, effect->desc().name, chain.get()); 1257 return BAD_VALUE; 1258 } 1259 1260 effect->setOffloaded(mType == OFFLOAD, mId); 1261 1262 status_t status = chain->addEffect_l(effect); 1263 if (status != NO_ERROR) { 1264 if (chainCreated) { 1265 removeEffectChain_l(chain); 1266 } 1267 return status; 1268 } 1269 1270 effect->setDevice(mOutDevice); 1271 effect->setDevice(mInDevice); 1272 effect->setMode(mAudioFlinger->getMode()); 1273 effect->setAudioSource(mAudioSource); 1274 return NO_ERROR; 1275} 1276 1277void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1278 1279 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1280 effect_descriptor_t desc = effect->desc(); 1281 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1282 detachAuxEffect_l(effect->id()); 1283 } 1284 1285 sp<EffectChain> chain = effect->chain().promote(); 1286 if (chain != 0) { 1287 // remove effect chain if removing last effect 1288 if (chain->removeEffect_l(effect) == 0) { 1289 removeEffectChain_l(chain); 1290 } 1291 } else { 1292 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1293 } 1294} 1295 1296void AudioFlinger::ThreadBase::lockEffectChains_l( 1297 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1298{ 1299 effectChains = mEffectChains; 1300 for (size_t i = 0; i < mEffectChains.size(); i++) { 1301 mEffectChains[i]->lock(); 1302 } 1303} 1304 1305void AudioFlinger::ThreadBase::unlockEffectChains( 1306 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1307{ 1308 for (size_t i = 0; i < effectChains.size(); i++) { 1309 effectChains[i]->unlock(); 1310 } 1311} 1312 1313sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1314{ 1315 Mutex::Autolock _l(mLock); 1316 return getEffectChain_l(sessionId); 1317} 1318 1319sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1320{ 1321 size_t size = mEffectChains.size(); 1322 for (size_t i = 0; i < size; i++) { 1323 if (mEffectChains[i]->sessionId() == sessionId) { 1324 return mEffectChains[i]; 1325 } 1326 } 1327 return 0; 1328} 1329 1330void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1331{ 1332 Mutex::Autolock _l(mLock); 1333 size_t size = mEffectChains.size(); 1334 for (size_t i = 0; i < size; i++) { 1335 mEffectChains[i]->setMode_l(mode); 1336 } 1337} 1338 1339void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1340{ 1341 config->type = AUDIO_PORT_TYPE_MIX; 1342 config->ext.mix.handle = mId; 1343 config->sample_rate = mSampleRate; 1344 config->format = mFormat; 1345 config->channel_mask = mChannelMask; 1346 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1347 AUDIO_PORT_CONFIG_FORMAT; 1348} 1349 1350void AudioFlinger::ThreadBase::systemReady() 1351{ 1352 Mutex::Autolock _l(mLock); 1353 if (mSystemReady) { 1354 return; 1355 } 1356 mSystemReady = true; 1357 1358 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1359 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1360 } 1361 mPendingConfigEvents.clear(); 1362} 1363 1364 1365// ---------------------------------------------------------------------------- 1366// Playback 1367// ---------------------------------------------------------------------------- 1368 1369AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1370 AudioStreamOut* output, 1371 audio_io_handle_t id, 1372 audio_devices_t device, 1373 type_t type, 1374 bool systemReady) 1375 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1376 mNormalFrameCount(0), mSinkBuffer(NULL), 1377 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1378 mMixerBuffer(NULL), 1379 mMixerBufferSize(0), 1380 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1381 mMixerBufferValid(false), 1382 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1383 mEffectBuffer(NULL), 1384 mEffectBufferSize(0), 1385 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1386 mEffectBufferValid(false), 1387 mSuspended(0), mBytesWritten(0), 1388 mActiveTracksGeneration(0), 1389 // mStreamTypes[] initialized in constructor body 1390 mOutput(output), 1391 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1392 mMixerStatus(MIXER_IDLE), 1393 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1394 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1395 mBytesRemaining(0), 1396 mCurrentWriteLength(0), 1397 mUseAsyncWrite(false), 1398 mWriteAckSequence(0), 1399 mDrainSequence(0), 1400 mSignalPending(false), 1401 mScreenState(AudioFlinger::mScreenState), 1402 // index 0 is reserved for normal mixer's submix 1403 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1404 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1405 // mLatchD, mLatchQ, 1406 mLatchDValid(false), mLatchQValid(false) 1407{ 1408 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1409 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1410 1411 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1412 // it would be safer to explicitly pass initial masterVolume/masterMute as 1413 // parameter. 1414 // 1415 // If the HAL we are using has support for master volume or master mute, 1416 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1417 // and the mute set to false). 1418 mMasterVolume = audioFlinger->masterVolume_l(); 1419 mMasterMute = audioFlinger->masterMute_l(); 1420 if (mOutput && mOutput->audioHwDev) { 1421 if (mOutput->audioHwDev->canSetMasterVolume()) { 1422 mMasterVolume = 1.0; 1423 } 1424 1425 if (mOutput->audioHwDev->canSetMasterMute()) { 1426 mMasterMute = false; 1427 } 1428 } 1429 1430 readOutputParameters_l(); 1431 1432 // ++ operator does not compile 1433 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1434 stream = (audio_stream_type_t) (stream + 1)) { 1435 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1436 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1437 } 1438} 1439 1440AudioFlinger::PlaybackThread::~PlaybackThread() 1441{ 1442 mAudioFlinger->unregisterWriter(mNBLogWriter); 1443 free(mSinkBuffer); 1444 free(mMixerBuffer); 1445 free(mEffectBuffer); 1446} 1447 1448void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1449{ 1450 dumpInternals(fd, args); 1451 dumpTracks(fd, args); 1452 dumpEffectChains(fd, args); 1453} 1454 1455void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1456{ 1457 const size_t SIZE = 256; 1458 char buffer[SIZE]; 1459 String8 result; 1460 1461 result.appendFormat(" Stream volumes in dB: "); 1462 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1463 const stream_type_t *st = &mStreamTypes[i]; 1464 if (i > 0) { 1465 result.appendFormat(", "); 1466 } 1467 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1468 if (st->mute) { 1469 result.append("M"); 1470 } 1471 } 1472 result.append("\n"); 1473 write(fd, result.string(), result.length()); 1474 result.clear(); 1475 1476 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1477 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1478 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1479 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1480 1481 size_t numtracks = mTracks.size(); 1482 size_t numactive = mActiveTracks.size(); 1483 dprintf(fd, " %d Tracks", numtracks); 1484 size_t numactiveseen = 0; 1485 if (numtracks) { 1486 dprintf(fd, " of which %d are active\n", numactive); 1487 Track::appendDumpHeader(result); 1488 for (size_t i = 0; i < numtracks; ++i) { 1489 sp<Track> track = mTracks[i]; 1490 if (track != 0) { 1491 bool active = mActiveTracks.indexOf(track) >= 0; 1492 if (active) { 1493 numactiveseen++; 1494 } 1495 track->dump(buffer, SIZE, active); 1496 result.append(buffer); 1497 } 1498 } 1499 } else { 1500 result.append("\n"); 1501 } 1502 if (numactiveseen != numactive) { 1503 // some tracks in the active list were not in the tracks list 1504 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1505 " not in the track list\n"); 1506 result.append(buffer); 1507 Track::appendDumpHeader(result); 1508 for (size_t i = 0; i < numactive; ++i) { 1509 sp<Track> track = mActiveTracks[i].promote(); 1510 if (track != 0 && mTracks.indexOf(track) < 0) { 1511 track->dump(buffer, SIZE, true); 1512 result.append(buffer); 1513 } 1514 } 1515 } 1516 1517 write(fd, result.string(), result.size()); 1518} 1519 1520void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1521{ 1522 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1523 1524 dumpBase(fd, args); 1525 1526 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1527 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1528 dprintf(fd, " Total writes: %d\n", mNumWrites); 1529 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1530 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1531 dprintf(fd, " Suspend count: %d\n", mSuspended); 1532 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1533 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1534 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1535 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1536 AudioStreamOut *output = mOutput; 1537 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1538 String8 flagsAsString = outputFlagsToString(flags); 1539 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1540} 1541 1542// Thread virtuals 1543 1544void AudioFlinger::PlaybackThread::onFirstRef() 1545{ 1546 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1547} 1548 1549// ThreadBase virtuals 1550void AudioFlinger::PlaybackThread::preExit() 1551{ 1552 ALOGV(" preExit()"); 1553 // FIXME this is using hard-coded strings but in the future, this functionality will be 1554 // converted to use audio HAL extensions required to support tunneling 1555 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1556} 1557 1558// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1559sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1560 const sp<AudioFlinger::Client>& client, 1561 audio_stream_type_t streamType, 1562 uint32_t sampleRate, 1563 audio_format_t format, 1564 audio_channel_mask_t channelMask, 1565 size_t *pFrameCount, 1566 const sp<IMemory>& sharedBuffer, 1567 int sessionId, 1568 IAudioFlinger::track_flags_t *flags, 1569 pid_t tid, 1570 int uid, 1571 status_t *status) 1572{ 1573 size_t frameCount = *pFrameCount; 1574 sp<Track> track; 1575 status_t lStatus; 1576 1577 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1578 1579 // client expresses a preference for FAST, but we get the final say 1580 if (*flags & IAudioFlinger::TRACK_FAST) { 1581 if ( 1582 // not timed 1583 (!isTimed) && 1584 // either of these use cases: 1585 ( 1586 // use case 1: shared buffer with any frame count 1587 ( 1588 (sharedBuffer != 0) 1589 ) || 1590 // use case 2: frame count is default or at least as large as HAL 1591 ( 1592 // we formerly checked for a callback handler (non-0 tid), 1593 // but that is no longer required for TRANSFER_OBTAIN mode 1594 ((frameCount == 0) || 1595 (frameCount >= mFrameCount)) 1596 ) 1597 ) && 1598 // PCM data 1599 audio_is_linear_pcm(format) && 1600 // identical channel mask to sink, or mono in and stereo sink 1601 (channelMask == mChannelMask || 1602 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1603 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1604 // hardware sample rate 1605 (sampleRate == mSampleRate) && 1606 // normal mixer has an associated fast mixer 1607 hasFastMixer() && 1608 // there are sufficient fast track slots available 1609 (mFastTrackAvailMask != 0) 1610 // FIXME test that MixerThread for this fast track has a capable output HAL 1611 // FIXME add a permission test also? 1612 ) { 1613 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1614 if (frameCount == 0) { 1615 // read the fast track multiplier property the first time it is needed 1616 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1617 if (ok != 0) { 1618 ALOGE("%s pthread_once failed: %d", __func__, ok); 1619 } 1620 frameCount = mFrameCount * sFastTrackMultiplier; 1621 } 1622 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1623 frameCount, mFrameCount); 1624 } else { 1625 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1626 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1627 "sampleRate=%u mSampleRate=%u " 1628 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1629 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1630 audio_is_linear_pcm(format), 1631 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1632 *flags &= ~IAudioFlinger::TRACK_FAST; 1633 } 1634 } 1635 // For normal PCM streaming tracks, update minimum frame count. 1636 // For compatibility with AudioTrack calculation, buffer depth is forced 1637 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1638 // This is probably too conservative, but legacy application code may depend on it. 1639 // If you change this calculation, also review the start threshold which is related. 1640 if (!(*flags & IAudioFlinger::TRACK_FAST) 1641 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1642 // this must match AudioTrack.cpp calculateMinFrameCount(). 1643 // TODO: Move to a common library 1644 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1645 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1646 if (minBufCount < 2) { 1647 minBufCount = 2; 1648 } 1649 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1650 // or the client should compute and pass in a larger buffer request. 1651 size_t minFrameCount = 1652 minBufCount * sourceFramesNeededWithTimestretch( 1653 sampleRate, mNormalFrameCount, 1654 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1655 if (frameCount < minFrameCount) { // including frameCount == 0 1656 frameCount = minFrameCount; 1657 } 1658 } 1659 *pFrameCount = frameCount; 1660 1661 switch (mType) { 1662 1663 case DIRECT: 1664 if (audio_is_linear_pcm(format)) { 1665 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1666 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1667 "for output %p with format %#x", 1668 sampleRate, format, channelMask, mOutput, mFormat); 1669 lStatus = BAD_VALUE; 1670 goto Exit; 1671 } 1672 } 1673 break; 1674 1675 case OFFLOAD: 1676 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1677 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1678 "for output %p with format %#x", 1679 sampleRate, format, channelMask, mOutput, mFormat); 1680 lStatus = BAD_VALUE; 1681 goto Exit; 1682 } 1683 break; 1684 1685 default: 1686 if (!audio_is_linear_pcm(format)) { 1687 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1688 "for output %p with format %#x", 1689 format, mOutput, mFormat); 1690 lStatus = BAD_VALUE; 1691 goto Exit; 1692 } 1693 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1694 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1695 lStatus = BAD_VALUE; 1696 goto Exit; 1697 } 1698 break; 1699 1700 } 1701 1702 lStatus = initCheck(); 1703 if (lStatus != NO_ERROR) { 1704 ALOGE("createTrack_l() audio driver not initialized"); 1705 goto Exit; 1706 } 1707 1708 { // scope for mLock 1709 Mutex::Autolock _l(mLock); 1710 1711 // all tracks in same audio session must share the same routing strategy otherwise 1712 // conflicts will happen when tracks are moved from one output to another by audio policy 1713 // manager 1714 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1715 for (size_t i = 0; i < mTracks.size(); ++i) { 1716 sp<Track> t = mTracks[i]; 1717 if (t != 0 && t->isExternalTrack()) { 1718 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1719 if (sessionId == t->sessionId() && strategy != actual) { 1720 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1721 strategy, actual); 1722 lStatus = BAD_VALUE; 1723 goto Exit; 1724 } 1725 } 1726 } 1727 1728 if (!isTimed) { 1729 track = new Track(this, client, streamType, sampleRate, format, 1730 channelMask, frameCount, NULL, sharedBuffer, 1731 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1732 } else { 1733 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1734 channelMask, frameCount, sharedBuffer, sessionId, uid); 1735 } 1736 1737 // new Track always returns non-NULL, 1738 // but TimedTrack::create() is a factory that could fail by returning NULL 1739 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1740 if (lStatus != NO_ERROR) { 1741 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1742 // track must be cleared from the caller as the caller has the AF lock 1743 goto Exit; 1744 } 1745 mTracks.add(track); 1746 1747 sp<EffectChain> chain = getEffectChain_l(sessionId); 1748 if (chain != 0) { 1749 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1750 track->setMainBuffer(chain->inBuffer()); 1751 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1752 chain->incTrackCnt(); 1753 } 1754 1755 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1756 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1757 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1758 // so ask activity manager to do this on our behalf 1759 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1760 } 1761 } 1762 1763 lStatus = NO_ERROR; 1764 1765Exit: 1766 *status = lStatus; 1767 return track; 1768} 1769 1770uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1771{ 1772 return latency; 1773} 1774 1775uint32_t AudioFlinger::PlaybackThread::latency() const 1776{ 1777 Mutex::Autolock _l(mLock); 1778 return latency_l(); 1779} 1780uint32_t AudioFlinger::PlaybackThread::latency_l() const 1781{ 1782 if (initCheck() == NO_ERROR) { 1783 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1784 } else { 1785 return 0; 1786 } 1787} 1788 1789void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1790{ 1791 Mutex::Autolock _l(mLock); 1792 // Don't apply master volume in SW if our HAL can do it for us. 1793 if (mOutput && mOutput->audioHwDev && 1794 mOutput->audioHwDev->canSetMasterVolume()) { 1795 mMasterVolume = 1.0; 1796 } else { 1797 mMasterVolume = value; 1798 } 1799} 1800 1801void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1802{ 1803 Mutex::Autolock _l(mLock); 1804 // Don't apply master mute in SW if our HAL can do it for us. 1805 if (mOutput && mOutput->audioHwDev && 1806 mOutput->audioHwDev->canSetMasterMute()) { 1807 mMasterMute = false; 1808 } else { 1809 mMasterMute = muted; 1810 } 1811} 1812 1813void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1814{ 1815 Mutex::Autolock _l(mLock); 1816 mStreamTypes[stream].volume = value; 1817 broadcast_l(); 1818} 1819 1820void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1821{ 1822 Mutex::Autolock _l(mLock); 1823 mStreamTypes[stream].mute = muted; 1824 broadcast_l(); 1825} 1826 1827float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1828{ 1829 Mutex::Autolock _l(mLock); 1830 return mStreamTypes[stream].volume; 1831} 1832 1833// addTrack_l() must be called with ThreadBase::mLock held 1834status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1835{ 1836 status_t status = ALREADY_EXISTS; 1837 1838 // set retry count for buffer fill 1839 track->mRetryCount = kMaxTrackStartupRetries; 1840 if (mActiveTracks.indexOf(track) < 0) { 1841 // the track is newly added, make sure it fills up all its 1842 // buffers before playing. This is to ensure the client will 1843 // effectively get the latency it requested. 1844 if (track->isExternalTrack()) { 1845 TrackBase::track_state state = track->mState; 1846 mLock.unlock(); 1847 status = AudioSystem::startOutput(mId, track->streamType(), 1848 (audio_session_t)track->sessionId()); 1849 mLock.lock(); 1850 // abort track was stopped/paused while we released the lock 1851 if (state != track->mState) { 1852 if (status == NO_ERROR) { 1853 mLock.unlock(); 1854 AudioSystem::stopOutput(mId, track->streamType(), 1855 (audio_session_t)track->sessionId()); 1856 mLock.lock(); 1857 } 1858 return INVALID_OPERATION; 1859 } 1860 // abort if start is rejected by audio policy manager 1861 if (status != NO_ERROR) { 1862 return PERMISSION_DENIED; 1863 } 1864#ifdef ADD_BATTERY_DATA 1865 // to track the speaker usage 1866 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1867#endif 1868 } 1869 1870 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1871 track->mResetDone = false; 1872 track->mPresentationCompleteFrames = 0; 1873 mActiveTracks.add(track); 1874 mWakeLockUids.add(track->uid()); 1875 mActiveTracksGeneration++; 1876 mLatestActiveTrack = track; 1877 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1878 if (chain != 0) { 1879 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1880 track->sessionId()); 1881 chain->incActiveTrackCnt(); 1882 } 1883 1884 status = NO_ERROR; 1885 } 1886 1887 onAddNewTrack_l(); 1888 return status; 1889} 1890 1891bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1892{ 1893 track->terminate(); 1894 // active tracks are removed by threadLoop() 1895 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1896 track->mState = TrackBase::STOPPED; 1897 if (!trackActive) { 1898 removeTrack_l(track); 1899 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1900 track->mState = TrackBase::STOPPING_1; 1901 } 1902 1903 return trackActive; 1904} 1905 1906void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1907{ 1908 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1909 mTracks.remove(track); 1910 deleteTrackName_l(track->name()); 1911 // redundant as track is about to be destroyed, for dumpsys only 1912 track->mName = -1; 1913 if (track->isFastTrack()) { 1914 int index = track->mFastIndex; 1915 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1916 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1917 mFastTrackAvailMask |= 1 << index; 1918 // redundant as track is about to be destroyed, for dumpsys only 1919 track->mFastIndex = -1; 1920 } 1921 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1922 if (chain != 0) { 1923 chain->decTrackCnt(); 1924 } 1925} 1926 1927void AudioFlinger::PlaybackThread::broadcast_l() 1928{ 1929 // Thread could be blocked waiting for async 1930 // so signal it to handle state changes immediately 1931 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1932 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1933 mSignalPending = true; 1934 mWaitWorkCV.broadcast(); 1935} 1936 1937String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1938{ 1939 Mutex::Autolock _l(mLock); 1940 if (initCheck() != NO_ERROR) { 1941 return String8(); 1942 } 1943 1944 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1945 const String8 out_s8(s); 1946 free(s); 1947 return out_s8; 1948} 1949 1950void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event) { 1951 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 1952 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 1953 1954 desc->mIoHandle = mId; 1955 1956 switch (event) { 1957 case AUDIO_OUTPUT_OPENED: 1958 case AUDIO_OUTPUT_CONFIG_CHANGED: 1959 desc->mPatch = mPatch; 1960 desc->mChannelMask = mChannelMask; 1961 desc->mSamplingRate = mSampleRate; 1962 desc->mFormat = mFormat; 1963 desc->mFrameCount = mNormalFrameCount; // FIXME see 1964 // AudioFlinger::frameCount(audio_io_handle_t) 1965 desc->mLatency = latency_l(); 1966 break; 1967 1968 case AUDIO_OUTPUT_CLOSED: 1969 default: 1970 break; 1971 } 1972 mAudioFlinger->ioConfigChanged(event, desc); 1973} 1974 1975void AudioFlinger::PlaybackThread::writeCallback() 1976{ 1977 ALOG_ASSERT(mCallbackThread != 0); 1978 mCallbackThread->resetWriteBlocked(); 1979} 1980 1981void AudioFlinger::PlaybackThread::drainCallback() 1982{ 1983 ALOG_ASSERT(mCallbackThread != 0); 1984 mCallbackThread->resetDraining(); 1985} 1986 1987void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1988{ 1989 Mutex::Autolock _l(mLock); 1990 // reject out of sequence requests 1991 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1992 mWriteAckSequence &= ~1; 1993 mWaitWorkCV.signal(); 1994 } 1995} 1996 1997void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1998{ 1999 Mutex::Autolock _l(mLock); 2000 // reject out of sequence requests 2001 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2002 mDrainSequence &= ~1; 2003 mWaitWorkCV.signal(); 2004 } 2005} 2006 2007// static 2008int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2009 void *param __unused, 2010 void *cookie) 2011{ 2012 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2013 ALOGV("asyncCallback() event %d", event); 2014 switch (event) { 2015 case STREAM_CBK_EVENT_WRITE_READY: 2016 me->writeCallback(); 2017 break; 2018 case STREAM_CBK_EVENT_DRAIN_READY: 2019 me->drainCallback(); 2020 break; 2021 default: 2022 ALOGW("asyncCallback() unknown event %d", event); 2023 break; 2024 } 2025 return 0; 2026} 2027 2028void AudioFlinger::PlaybackThread::readOutputParameters_l() 2029{ 2030 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2031 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2032 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2033 if (!audio_is_output_channel(mChannelMask)) { 2034 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2035 } 2036 if ((mType == MIXER || mType == DUPLICATING) 2037 && !isValidPcmSinkChannelMask(mChannelMask)) { 2038 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2039 mChannelMask); 2040 } 2041 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2042 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2043 mFormat = mHALFormat; 2044 if (!audio_is_valid_format(mFormat)) { 2045 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2046 } 2047 if ((mType == MIXER || mType == DUPLICATING) 2048 && !isValidPcmSinkFormat(mFormat)) { 2049 LOG_FATAL("HAL format %#x not supported for mixed output", 2050 mFormat); 2051 } 2052 mFrameSize = mOutput->getFrameSize(); 2053 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2054 mFrameCount = mBufferSize / mFrameSize; 2055 if (mFrameCount & 15) { 2056 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2057 mFrameCount); 2058 } 2059 2060 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2061 (mOutput->stream->set_callback != NULL)) { 2062 if (mOutput->stream->set_callback(mOutput->stream, 2063 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2064 mUseAsyncWrite = true; 2065 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2066 } 2067 } 2068 2069 mHwSupportsPause = false; 2070 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2071 if (mOutput->stream->pause != NULL) { 2072 if (mOutput->stream->resume != NULL) { 2073 mHwSupportsPause = true; 2074 } else { 2075 ALOGW("direct output implements pause but not resume"); 2076 } 2077 } else if (mOutput->stream->resume != NULL) { 2078 ALOGW("direct output implements resume but not pause"); 2079 } 2080 } 2081 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2082 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2083 } 2084 2085 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2086 // For best precision, we use float instead of the associated output 2087 // device format (typically PCM 16 bit). 2088 2089 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2090 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2091 mBufferSize = mFrameSize * mFrameCount; 2092 2093 // TODO: We currently use the associated output device channel mask and sample rate. 2094 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2095 // (if a valid mask) to avoid premature downmix. 2096 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2097 // instead of the output device sample rate to avoid loss of high frequency information. 2098 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2099 } 2100 2101 // Calculate size of normal sink buffer relative to the HAL output buffer size 2102 double multiplier = 1.0; 2103 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2104 kUseFastMixer == FastMixer_Dynamic)) { 2105 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2106 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2107 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2108 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2109 maxNormalFrameCount = maxNormalFrameCount & ~15; 2110 if (maxNormalFrameCount < minNormalFrameCount) { 2111 maxNormalFrameCount = minNormalFrameCount; 2112 } 2113 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2114 if (multiplier <= 1.0) { 2115 multiplier = 1.0; 2116 } else if (multiplier <= 2.0) { 2117 if (2 * mFrameCount <= maxNormalFrameCount) { 2118 multiplier = 2.0; 2119 } else { 2120 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2121 } 2122 } else { 2123 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2124 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2125 // track, but we sometimes have to do this to satisfy the maximum frame count 2126 // constraint) 2127 // FIXME this rounding up should not be done if no HAL SRC 2128 uint32_t truncMult = (uint32_t) multiplier; 2129 if ((truncMult & 1)) { 2130 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2131 ++truncMult; 2132 } 2133 } 2134 multiplier = (double) truncMult; 2135 } 2136 } 2137 mNormalFrameCount = multiplier * mFrameCount; 2138 // round up to nearest 16 frames to satisfy AudioMixer 2139 if (mType == MIXER || mType == DUPLICATING) { 2140 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2141 } 2142 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2143 mNormalFrameCount); 2144 2145 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2146 // Originally this was int16_t[] array, need to remove legacy implications. 2147 free(mSinkBuffer); 2148 mSinkBuffer = NULL; 2149 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2150 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2151 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2152 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2153 2154 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2155 // drives the output. 2156 free(mMixerBuffer); 2157 mMixerBuffer = NULL; 2158 if (mMixerBufferEnabled) { 2159 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2160 mMixerBufferSize = mNormalFrameCount * mChannelCount 2161 * audio_bytes_per_sample(mMixerBufferFormat); 2162 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2163 } 2164 free(mEffectBuffer); 2165 mEffectBuffer = NULL; 2166 if (mEffectBufferEnabled) { 2167 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2168 mEffectBufferSize = mNormalFrameCount * mChannelCount 2169 * audio_bytes_per_sample(mEffectBufferFormat); 2170 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2171 } 2172 2173 // force reconfiguration of effect chains and engines to take new buffer size and audio 2174 // parameters into account 2175 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2176 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2177 // matter. 2178 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2179 Vector< sp<EffectChain> > effectChains = mEffectChains; 2180 for (size_t i = 0; i < effectChains.size(); i ++) { 2181 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2182 } 2183} 2184 2185 2186status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2187{ 2188 if (halFrames == NULL || dspFrames == NULL) { 2189 return BAD_VALUE; 2190 } 2191 Mutex::Autolock _l(mLock); 2192 if (initCheck() != NO_ERROR) { 2193 return INVALID_OPERATION; 2194 } 2195 size_t framesWritten = mBytesWritten / mFrameSize; 2196 *halFrames = framesWritten; 2197 2198 if (isSuspended()) { 2199 // return an estimation of rendered frames when the output is suspended 2200 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2201 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2202 return NO_ERROR; 2203 } else { 2204 status_t status; 2205 uint32_t frames; 2206 status = mOutput->getRenderPosition(&frames); 2207 *dspFrames = (size_t)frames; 2208 return status; 2209 } 2210} 2211 2212uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2213{ 2214 Mutex::Autolock _l(mLock); 2215 uint32_t result = 0; 2216 if (getEffectChain_l(sessionId) != 0) { 2217 result = EFFECT_SESSION; 2218 } 2219 2220 for (size_t i = 0; i < mTracks.size(); ++i) { 2221 sp<Track> track = mTracks[i]; 2222 if (sessionId == track->sessionId() && !track->isInvalid()) { 2223 result |= TRACK_SESSION; 2224 break; 2225 } 2226 } 2227 2228 return result; 2229} 2230 2231uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2232{ 2233 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2234 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2235 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2236 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2237 } 2238 for (size_t i = 0; i < mTracks.size(); i++) { 2239 sp<Track> track = mTracks[i]; 2240 if (sessionId == track->sessionId() && !track->isInvalid()) { 2241 return AudioSystem::getStrategyForStream(track->streamType()); 2242 } 2243 } 2244 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2245} 2246 2247 2248AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2249{ 2250 Mutex::Autolock _l(mLock); 2251 return mOutput; 2252} 2253 2254AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2255{ 2256 Mutex::Autolock _l(mLock); 2257 AudioStreamOut *output = mOutput; 2258 mOutput = NULL; 2259 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2260 // must push a NULL and wait for ack 2261 mOutputSink.clear(); 2262 mPipeSink.clear(); 2263 mNormalSink.clear(); 2264 return output; 2265} 2266 2267// this method must always be called either with ThreadBase mLock held or inside the thread loop 2268audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2269{ 2270 if (mOutput == NULL) { 2271 return NULL; 2272 } 2273 return &mOutput->stream->common; 2274} 2275 2276uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2277{ 2278 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2279} 2280 2281status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2282{ 2283 if (!isValidSyncEvent(event)) { 2284 return BAD_VALUE; 2285 } 2286 2287 Mutex::Autolock _l(mLock); 2288 2289 for (size_t i = 0; i < mTracks.size(); ++i) { 2290 sp<Track> track = mTracks[i]; 2291 if (event->triggerSession() == track->sessionId()) { 2292 (void) track->setSyncEvent(event); 2293 return NO_ERROR; 2294 } 2295 } 2296 2297 return NAME_NOT_FOUND; 2298} 2299 2300bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2301{ 2302 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2303} 2304 2305void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2306 const Vector< sp<Track> >& tracksToRemove) 2307{ 2308 size_t count = tracksToRemove.size(); 2309 if (count > 0) { 2310 for (size_t i = 0 ; i < count ; i++) { 2311 const sp<Track>& track = tracksToRemove.itemAt(i); 2312 if (track->isExternalTrack()) { 2313 AudioSystem::stopOutput(mId, track->streamType(), 2314 (audio_session_t)track->sessionId()); 2315#ifdef ADD_BATTERY_DATA 2316 // to track the speaker usage 2317 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2318#endif 2319 if (track->isTerminated()) { 2320 AudioSystem::releaseOutput(mId, track->streamType(), 2321 (audio_session_t)track->sessionId()); 2322 } 2323 } 2324 } 2325 } 2326} 2327 2328void AudioFlinger::PlaybackThread::checkSilentMode_l() 2329{ 2330 if (!mMasterMute) { 2331 char value[PROPERTY_VALUE_MAX]; 2332 if (property_get("ro.audio.silent", value, "0") > 0) { 2333 char *endptr; 2334 unsigned long ul = strtoul(value, &endptr, 0); 2335 if (*endptr == '\0' && ul != 0) { 2336 ALOGD("Silence is golden"); 2337 // The setprop command will not allow a property to be changed after 2338 // the first time it is set, so we don't have to worry about un-muting. 2339 setMasterMute_l(true); 2340 } 2341 } 2342 } 2343} 2344 2345// shared by MIXER and DIRECT, overridden by DUPLICATING 2346ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2347{ 2348 // FIXME rewrite to reduce number of system calls 2349 mLastWriteTime = systemTime(); 2350 mInWrite = true; 2351 ssize_t bytesWritten; 2352 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2353 2354 // If an NBAIO sink is present, use it to write the normal mixer's submix 2355 if (mNormalSink != 0) { 2356 2357 const size_t count = mBytesRemaining / mFrameSize; 2358 2359 ATRACE_BEGIN("write"); 2360 // update the setpoint when AudioFlinger::mScreenState changes 2361 uint32_t screenState = AudioFlinger::mScreenState; 2362 if (screenState != mScreenState) { 2363 mScreenState = screenState; 2364 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2365 if (pipe != NULL) { 2366 pipe->setAvgFrames((mScreenState & 1) ? 2367 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2368 } 2369 } 2370 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2371 ATRACE_END(); 2372 if (framesWritten > 0) { 2373 bytesWritten = framesWritten * mFrameSize; 2374 } else { 2375 bytesWritten = framesWritten; 2376 } 2377 mLatchDValid = false; 2378 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2379 if (status == NO_ERROR) { 2380 size_t totalFramesWritten = mNormalSink->framesWritten(); 2381 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2382 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2383 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2384 mLatchDValid = true; 2385 } 2386 } 2387 // otherwise use the HAL / AudioStreamOut directly 2388 } else { 2389 // Direct output and offload threads 2390 2391 if (mUseAsyncWrite) { 2392 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2393 mWriteAckSequence += 2; 2394 mWriteAckSequence |= 1; 2395 ALOG_ASSERT(mCallbackThread != 0); 2396 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2397 } 2398 // FIXME We should have an implementation of timestamps for direct output threads. 2399 // They are used e.g for multichannel PCM playback over HDMI. 2400 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2401 if (mUseAsyncWrite && 2402 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2403 // do not wait for async callback in case of error of full write 2404 mWriteAckSequence &= ~1; 2405 ALOG_ASSERT(mCallbackThread != 0); 2406 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2407 } 2408 } 2409 2410 mNumWrites++; 2411 mInWrite = false; 2412 mStandby = false; 2413 return bytesWritten; 2414} 2415 2416void AudioFlinger::PlaybackThread::threadLoop_drain() 2417{ 2418 if (mOutput->stream->drain) { 2419 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2420 if (mUseAsyncWrite) { 2421 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2422 mDrainSequence |= 1; 2423 ALOG_ASSERT(mCallbackThread != 0); 2424 mCallbackThread->setDraining(mDrainSequence); 2425 } 2426 mOutput->stream->drain(mOutput->stream, 2427 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2428 : AUDIO_DRAIN_ALL); 2429 } 2430} 2431 2432void AudioFlinger::PlaybackThread::threadLoop_exit() 2433{ 2434 { 2435 Mutex::Autolock _l(mLock); 2436 for (size_t i = 0; i < mTracks.size(); i++) { 2437 sp<Track> track = mTracks[i]; 2438 track->invalidate(); 2439 } 2440 } 2441} 2442 2443/* 2444The derived values that are cached: 2445 - mSinkBufferSize from frame count * frame size 2446 - activeSleepTime from activeSleepTimeUs() 2447 - idleSleepTime from idleSleepTimeUs() 2448 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2449 - maxPeriod from frame count and sample rate (MIXER only) 2450 2451The parameters that affect these derived values are: 2452 - frame count 2453 - frame size 2454 - sample rate 2455 - device type: A2DP or not 2456 - device latency 2457 - format: PCM or not 2458 - active sleep time 2459 - idle sleep time 2460*/ 2461 2462void AudioFlinger::PlaybackThread::cacheParameters_l() 2463{ 2464 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2465 activeSleepTime = activeSleepTimeUs(); 2466 idleSleepTime = idleSleepTimeUs(); 2467} 2468 2469void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2470{ 2471 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2472 this, streamType, mTracks.size()); 2473 Mutex::Autolock _l(mLock); 2474 2475 size_t size = mTracks.size(); 2476 for (size_t i = 0; i < size; i++) { 2477 sp<Track> t = mTracks[i]; 2478 if (t->streamType() == streamType) { 2479 t->invalidate(); 2480 } 2481 } 2482} 2483 2484status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2485{ 2486 int session = chain->sessionId(); 2487 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2488 ? mEffectBuffer : mSinkBuffer); 2489 bool ownsBuffer = false; 2490 2491 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2492 if (session > 0) { 2493 // Only one effect chain can be present in direct output thread and it uses 2494 // the sink buffer as input 2495 if (mType != DIRECT) { 2496 size_t numSamples = mNormalFrameCount * mChannelCount; 2497 buffer = new int16_t[numSamples]; 2498 memset(buffer, 0, numSamples * sizeof(int16_t)); 2499 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2500 ownsBuffer = true; 2501 } 2502 2503 // Attach all tracks with same session ID to this chain. 2504 for (size_t i = 0; i < mTracks.size(); ++i) { 2505 sp<Track> track = mTracks[i]; 2506 if (session == track->sessionId()) { 2507 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2508 buffer); 2509 track->setMainBuffer(buffer); 2510 chain->incTrackCnt(); 2511 } 2512 } 2513 2514 // indicate all active tracks in the chain 2515 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2516 sp<Track> track = mActiveTracks[i].promote(); 2517 if (track == 0) { 2518 continue; 2519 } 2520 if (session == track->sessionId()) { 2521 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2522 chain->incActiveTrackCnt(); 2523 } 2524 } 2525 } 2526 chain->setThread(this); 2527 chain->setInBuffer(buffer, ownsBuffer); 2528 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2529 ? mEffectBuffer : mSinkBuffer)); 2530 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2531 // chains list in order to be processed last as it contains output stage effects 2532 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2533 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2534 // after track specific effects and before output stage 2535 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2536 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2537 // Effect chain for other sessions are inserted at beginning of effect 2538 // chains list to be processed before output mix effects. Relative order between other 2539 // sessions is not important 2540 size_t size = mEffectChains.size(); 2541 size_t i = 0; 2542 for (i = 0; i < size; i++) { 2543 if (mEffectChains[i]->sessionId() < session) { 2544 break; 2545 } 2546 } 2547 mEffectChains.insertAt(chain, i); 2548 checkSuspendOnAddEffectChain_l(chain); 2549 2550 return NO_ERROR; 2551} 2552 2553size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2554{ 2555 int session = chain->sessionId(); 2556 2557 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2558 2559 for (size_t i = 0; i < mEffectChains.size(); i++) { 2560 if (chain == mEffectChains[i]) { 2561 mEffectChains.removeAt(i); 2562 // detach all active tracks from the chain 2563 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2564 sp<Track> track = mActiveTracks[i].promote(); 2565 if (track == 0) { 2566 continue; 2567 } 2568 if (session == track->sessionId()) { 2569 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2570 chain.get(), session); 2571 chain->decActiveTrackCnt(); 2572 } 2573 } 2574 2575 // detach all tracks with same session ID from this chain 2576 for (size_t i = 0; i < mTracks.size(); ++i) { 2577 sp<Track> track = mTracks[i]; 2578 if (session == track->sessionId()) { 2579 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2580 chain->decTrackCnt(); 2581 } 2582 } 2583 break; 2584 } 2585 } 2586 return mEffectChains.size(); 2587} 2588 2589status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2590 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2591{ 2592 Mutex::Autolock _l(mLock); 2593 return attachAuxEffect_l(track, EffectId); 2594} 2595 2596status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2597 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2598{ 2599 status_t status = NO_ERROR; 2600 2601 if (EffectId == 0) { 2602 track->setAuxBuffer(0, NULL); 2603 } else { 2604 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2605 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2606 if (effect != 0) { 2607 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2608 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2609 } else { 2610 status = INVALID_OPERATION; 2611 } 2612 } else { 2613 status = BAD_VALUE; 2614 } 2615 } 2616 return status; 2617} 2618 2619void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2620{ 2621 for (size_t i = 0; i < mTracks.size(); ++i) { 2622 sp<Track> track = mTracks[i]; 2623 if (track->auxEffectId() == effectId) { 2624 attachAuxEffect_l(track, 0); 2625 } 2626 } 2627} 2628 2629bool AudioFlinger::PlaybackThread::threadLoop() 2630{ 2631 Vector< sp<Track> > tracksToRemove; 2632 2633 standbyTime = systemTime(); 2634 2635 // MIXER 2636 nsecs_t lastWarning = 0; 2637 2638 // DUPLICATING 2639 // FIXME could this be made local to while loop? 2640 writeFrames = 0; 2641 2642 int lastGeneration = 0; 2643 2644 cacheParameters_l(); 2645 sleepTime = idleSleepTime; 2646 2647 if (mType == MIXER) { 2648 sleepTimeShift = 0; 2649 } 2650 2651 CpuStats cpuStats; 2652 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2653 2654 acquireWakeLock(); 2655 2656 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2657 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2658 // and then that string will be logged at the next convenient opportunity. 2659 const char *logString = NULL; 2660 2661 checkSilentMode_l(); 2662 2663 while (!exitPending()) 2664 { 2665 cpuStats.sample(myName); 2666 2667 Vector< sp<EffectChain> > effectChains; 2668 2669 { // scope for mLock 2670 2671 Mutex::Autolock _l(mLock); 2672 2673 processConfigEvents_l(); 2674 2675 if (logString != NULL) { 2676 mNBLogWriter->logTimestamp(); 2677 mNBLogWriter->log(logString); 2678 logString = NULL; 2679 } 2680 2681 // Gather the framesReleased counters for all active tracks, 2682 // and latch them atomically with the timestamp. 2683 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2684 mLatchD.mFramesReleased.clear(); 2685 size_t size = mActiveTracks.size(); 2686 for (size_t i = 0; i < size; i++) { 2687 sp<Track> t = mActiveTracks[i].promote(); 2688 if (t != 0) { 2689 mLatchD.mFramesReleased.add(t.get(), 2690 t->mAudioTrackServerProxy->framesReleased()); 2691 } 2692 } 2693 if (mLatchDValid) { 2694 mLatchQ = mLatchD; 2695 mLatchDValid = false; 2696 mLatchQValid = true; 2697 } 2698 2699 saveOutputTracks(); 2700 if (mSignalPending) { 2701 // A signal was raised while we were unlocked 2702 mSignalPending = false; 2703 } else if (waitingAsyncCallback_l()) { 2704 if (exitPending()) { 2705 break; 2706 } 2707 bool released = false; 2708 // The following works around a bug in the offload driver. Ideally we would release 2709 // the wake lock every time, but that causes the last offload buffer(s) to be 2710 // dropped while the device is on battery, so we need to hold a wake lock during 2711 // the drain phase. 2712 if (mBytesRemaining && !(mDrainSequence & 1)) { 2713 releaseWakeLock_l(); 2714 released = true; 2715 } 2716 mWakeLockUids.clear(); 2717 mActiveTracksGeneration++; 2718 ALOGV("wait async completion"); 2719 mWaitWorkCV.wait(mLock); 2720 ALOGV("async completion/wake"); 2721 if (released) { 2722 acquireWakeLock_l(); 2723 } 2724 standbyTime = systemTime() + standbyDelay; 2725 sleepTime = 0; 2726 2727 continue; 2728 } 2729 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2730 isSuspended()) { 2731 // put audio hardware into standby after short delay 2732 if (shouldStandby_l()) { 2733 2734 threadLoop_standby(); 2735 2736 mStandby = true; 2737 } 2738 2739 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2740 // we're about to wait, flush the binder command buffer 2741 IPCThreadState::self()->flushCommands(); 2742 2743 clearOutputTracks(); 2744 2745 if (exitPending()) { 2746 break; 2747 } 2748 2749 releaseWakeLock_l(); 2750 mWakeLockUids.clear(); 2751 mActiveTracksGeneration++; 2752 // wait until we have something to do... 2753 ALOGV("%s going to sleep", myName.string()); 2754 mWaitWorkCV.wait(mLock); 2755 ALOGV("%s waking up", myName.string()); 2756 acquireWakeLock_l(); 2757 2758 mMixerStatus = MIXER_IDLE; 2759 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2760 mBytesWritten = 0; 2761 mBytesRemaining = 0; 2762 checkSilentMode_l(); 2763 2764 standbyTime = systemTime() + standbyDelay; 2765 sleepTime = idleSleepTime; 2766 if (mType == MIXER) { 2767 sleepTimeShift = 0; 2768 } 2769 2770 continue; 2771 } 2772 } 2773 // mMixerStatusIgnoringFastTracks is also updated internally 2774 mMixerStatus = prepareTracks_l(&tracksToRemove); 2775 2776 // compare with previously applied list 2777 if (lastGeneration != mActiveTracksGeneration) { 2778 // update wakelock 2779 updateWakeLockUids_l(mWakeLockUids); 2780 lastGeneration = mActiveTracksGeneration; 2781 } 2782 2783 // prevent any changes in effect chain list and in each effect chain 2784 // during mixing and effect process as the audio buffers could be deleted 2785 // or modified if an effect is created or deleted 2786 lockEffectChains_l(effectChains); 2787 } // mLock scope ends 2788 2789 if (mBytesRemaining == 0) { 2790 mCurrentWriteLength = 0; 2791 if (mMixerStatus == MIXER_TRACKS_READY) { 2792 // threadLoop_mix() sets mCurrentWriteLength 2793 threadLoop_mix(); 2794 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2795 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2796 // threadLoop_sleepTime sets sleepTime to 0 if data 2797 // must be written to HAL 2798 threadLoop_sleepTime(); 2799 if (sleepTime == 0) { 2800 mCurrentWriteLength = mSinkBufferSize; 2801 } 2802 } 2803 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2804 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2805 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2806 // or mSinkBuffer (if there are no effects). 2807 // 2808 // This is done pre-effects computation; if effects change to 2809 // support higher precision, this needs to move. 2810 // 2811 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2812 // TODO use sleepTime == 0 as an additional condition. 2813 if (mMixerBufferValid) { 2814 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2815 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2816 2817 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2818 mNormalFrameCount * mChannelCount); 2819 } 2820 2821 mBytesRemaining = mCurrentWriteLength; 2822 if (isSuspended()) { 2823 sleepTime = suspendSleepTimeUs(); 2824 // simulate write to HAL when suspended 2825 mBytesWritten += mSinkBufferSize; 2826 mBytesRemaining = 0; 2827 } 2828 2829 // only process effects if we're going to write 2830 if (sleepTime == 0 && mType != OFFLOAD) { 2831 for (size_t i = 0; i < effectChains.size(); i ++) { 2832 effectChains[i]->process_l(); 2833 } 2834 } 2835 } 2836 // Process effect chains for offloaded thread even if no audio 2837 // was read from audio track: process only updates effect state 2838 // and thus does have to be synchronized with audio writes but may have 2839 // to be called while waiting for async write callback 2840 if (mType == OFFLOAD) { 2841 for (size_t i = 0; i < effectChains.size(); i ++) { 2842 effectChains[i]->process_l(); 2843 } 2844 } 2845 2846 // Only if the Effects buffer is enabled and there is data in the 2847 // Effects buffer (buffer valid), we need to 2848 // copy into the sink buffer. 2849 // TODO use sleepTime == 0 as an additional condition. 2850 if (mEffectBufferValid) { 2851 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2852 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2853 mNormalFrameCount * mChannelCount); 2854 } 2855 2856 // enable changes in effect chain 2857 unlockEffectChains(effectChains); 2858 2859 if (!waitingAsyncCallback()) { 2860 // sleepTime == 0 means we must write to audio hardware 2861 if (sleepTime == 0) { 2862 if (mBytesRemaining) { 2863 ssize_t ret = threadLoop_write(); 2864 if (ret < 0) { 2865 mBytesRemaining = 0; 2866 } else { 2867 mBytesWritten += ret; 2868 mBytesRemaining -= ret; 2869 } 2870 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2871 (mMixerStatus == MIXER_DRAIN_ALL)) { 2872 threadLoop_drain(); 2873 } 2874 if (mType == MIXER) { 2875 // write blocked detection 2876 nsecs_t now = systemTime(); 2877 nsecs_t delta = now - mLastWriteTime; 2878 if (!mStandby && delta > maxPeriod) { 2879 mNumDelayedWrites++; 2880 if ((now - lastWarning) > kWarningThrottleNs) { 2881 ATRACE_NAME("underrun"); 2882 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2883 ns2ms(delta), mNumDelayedWrites, this); 2884 lastWarning = now; 2885 } 2886 } 2887 } 2888 2889 } else { 2890 ATRACE_BEGIN("sleep"); 2891 usleep(sleepTime); 2892 ATRACE_END(); 2893 } 2894 } 2895 2896 // Finally let go of removed track(s), without the lock held 2897 // since we can't guarantee the destructors won't acquire that 2898 // same lock. This will also mutate and push a new fast mixer state. 2899 threadLoop_removeTracks(tracksToRemove); 2900 tracksToRemove.clear(); 2901 2902 // FIXME I don't understand the need for this here; 2903 // it was in the original code but maybe the 2904 // assignment in saveOutputTracks() makes this unnecessary? 2905 clearOutputTracks(); 2906 2907 // Effect chains will be actually deleted here if they were removed from 2908 // mEffectChains list during mixing or effects processing 2909 effectChains.clear(); 2910 2911 // FIXME Note that the above .clear() is no longer necessary since effectChains 2912 // is now local to this block, but will keep it for now (at least until merge done). 2913 } 2914 2915 threadLoop_exit(); 2916 2917 if (!mStandby) { 2918 threadLoop_standby(); 2919 mStandby = true; 2920 } 2921 2922 releaseWakeLock(); 2923 mWakeLockUids.clear(); 2924 mActiveTracksGeneration++; 2925 2926 ALOGV("Thread %p type %d exiting", this, mType); 2927 return false; 2928} 2929 2930// removeTracks_l() must be called with ThreadBase::mLock held 2931void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2932{ 2933 size_t count = tracksToRemove.size(); 2934 if (count > 0) { 2935 for (size_t i=0 ; i<count ; i++) { 2936 const sp<Track>& track = tracksToRemove.itemAt(i); 2937 mActiveTracks.remove(track); 2938 mWakeLockUids.remove(track->uid()); 2939 mActiveTracksGeneration++; 2940 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2941 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2942 if (chain != 0) { 2943 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2944 track->sessionId()); 2945 chain->decActiveTrackCnt(); 2946 } 2947 if (track->isTerminated()) { 2948 removeTrack_l(track); 2949 } 2950 } 2951 } 2952 2953} 2954 2955status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2956{ 2957 if (mNormalSink != 0) { 2958 return mNormalSink->getTimestamp(timestamp); 2959 } 2960 if ((mType == OFFLOAD || mType == DIRECT) 2961 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2962 uint64_t position64; 2963 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 2964 if (ret == 0) { 2965 timestamp.mPosition = (uint32_t)position64; 2966 return NO_ERROR; 2967 } 2968 } 2969 return INVALID_OPERATION; 2970} 2971 2972status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 2973 audio_patch_handle_t *handle) 2974{ 2975 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2976 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2977 if (mFastMixer != 0) { 2978 FastMixerStateQueue *sq = mFastMixer->sq(); 2979 FastMixerState *state = sq->begin(); 2980 if (!(state->mCommand & FastMixerState::IDLE)) { 2981 previousCommand = state->mCommand; 2982 state->mCommand = FastMixerState::HOT_IDLE; 2983 sq->end(); 2984 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2985 } else { 2986 sq->end(false /*didModify*/); 2987 } 2988 } 2989 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 2990 2991 if (!(previousCommand & FastMixerState::IDLE)) { 2992 ALOG_ASSERT(mFastMixer != 0); 2993 FastMixerStateQueue *sq = mFastMixer->sq(); 2994 FastMixerState *state = sq->begin(); 2995 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 2996 state->mCommand = previousCommand; 2997 sq->end(); 2998 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2999 } 3000 3001 return status; 3002} 3003 3004status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3005 audio_patch_handle_t *handle) 3006{ 3007 status_t status = NO_ERROR; 3008 3009 // store new device and send to effects 3010 audio_devices_t type = AUDIO_DEVICE_NONE; 3011 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3012 type |= patch->sinks[i].ext.device.type; 3013 } 3014 3015#ifdef ADD_BATTERY_DATA 3016 // when changing the audio output device, call addBatteryData to notify 3017 // the change 3018 if (mOutDevice != type) { 3019 uint32_t params = 0; 3020 // check whether speaker is on 3021 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3022 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3023 } 3024 3025 audio_devices_t deviceWithoutSpeaker 3026 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3027 // check if any other device (except speaker) is on 3028 if (type & deviceWithoutSpeaker) { 3029 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3030 } 3031 3032 if (params != 0) { 3033 addBatteryData(params); 3034 } 3035 } 3036#endif 3037 3038 for (size_t i = 0; i < mEffectChains.size(); i++) { 3039 mEffectChains[i]->setDevice_l(type); 3040 } 3041 mOutDevice = type; 3042 mPatch = *patch; 3043 3044 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3045 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3046 status = hwDevice->create_audio_patch(hwDevice, 3047 patch->num_sources, 3048 patch->sources, 3049 patch->num_sinks, 3050 patch->sinks, 3051 handle); 3052 } else { 3053 char *address; 3054 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3055 //FIXME: we only support address on first sink with HAL version < 3.0 3056 address = audio_device_address_to_parameter( 3057 patch->sinks[0].ext.device.type, 3058 patch->sinks[0].ext.device.address); 3059 } else { 3060 address = (char *)calloc(1, 1); 3061 } 3062 AudioParameter param = AudioParameter(String8(address)); 3063 free(address); 3064 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3065 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3066 param.toString().string()); 3067 *handle = AUDIO_PATCH_HANDLE_NONE; 3068 } 3069 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3070 return status; 3071} 3072 3073status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3074{ 3075 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3076 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3077 if (mFastMixer != 0) { 3078 FastMixerStateQueue *sq = mFastMixer->sq(); 3079 FastMixerState *state = sq->begin(); 3080 if (!(state->mCommand & FastMixerState::IDLE)) { 3081 previousCommand = state->mCommand; 3082 state->mCommand = FastMixerState::HOT_IDLE; 3083 sq->end(); 3084 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3085 } else { 3086 sq->end(false /*didModify*/); 3087 } 3088 } 3089 3090 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3091 3092 if (!(previousCommand & FastMixerState::IDLE)) { 3093 ALOG_ASSERT(mFastMixer != 0); 3094 FastMixerStateQueue *sq = mFastMixer->sq(); 3095 FastMixerState *state = sq->begin(); 3096 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3097 state->mCommand = previousCommand; 3098 sq->end(); 3099 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3100 } 3101 3102 return status; 3103} 3104 3105status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3106{ 3107 status_t status = NO_ERROR; 3108 3109 mOutDevice = AUDIO_DEVICE_NONE; 3110 3111 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3112 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3113 status = hwDevice->release_audio_patch(hwDevice, handle); 3114 } else { 3115 AudioParameter param; 3116 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3117 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3118 param.toString().string()); 3119 } 3120 return status; 3121} 3122 3123void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3124{ 3125 Mutex::Autolock _l(mLock); 3126 mTracks.add(track); 3127} 3128 3129void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3130{ 3131 Mutex::Autolock _l(mLock); 3132 destroyTrack_l(track); 3133} 3134 3135void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3136{ 3137 ThreadBase::getAudioPortConfig(config); 3138 config->role = AUDIO_PORT_ROLE_SOURCE; 3139 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3140 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3141} 3142 3143// ---------------------------------------------------------------------------- 3144 3145AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3146 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3147 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3148 // mAudioMixer below 3149 // mFastMixer below 3150 mFastMixerFutex(0) 3151 // mOutputSink below 3152 // mPipeSink below 3153 // mNormalSink below 3154{ 3155 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3156 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3157 "mFrameCount=%d, mNormalFrameCount=%d", 3158 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3159 mNormalFrameCount); 3160 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3161 3162 if (type == DUPLICATING) { 3163 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3164 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3165 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3166 return; 3167 } 3168 // create an NBAIO sink for the HAL output stream, and negotiate 3169 mOutputSink = new AudioStreamOutSink(output->stream); 3170 size_t numCounterOffers = 0; 3171 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3172 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3173 ALOG_ASSERT(index == 0); 3174 3175 // initialize fast mixer depending on configuration 3176 bool initFastMixer; 3177 switch (kUseFastMixer) { 3178 case FastMixer_Never: 3179 initFastMixer = false; 3180 break; 3181 case FastMixer_Always: 3182 initFastMixer = true; 3183 break; 3184 case FastMixer_Static: 3185 case FastMixer_Dynamic: 3186 initFastMixer = mFrameCount < mNormalFrameCount; 3187 break; 3188 } 3189 if (initFastMixer) { 3190 audio_format_t fastMixerFormat; 3191 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3192 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3193 } else { 3194 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3195 } 3196 if (mFormat != fastMixerFormat) { 3197 // change our Sink format to accept our intermediate precision 3198 mFormat = fastMixerFormat; 3199 free(mSinkBuffer); 3200 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3201 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3202 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3203 } 3204 3205 // create a MonoPipe to connect our submix to FastMixer 3206 NBAIO_Format format = mOutputSink->format(); 3207 NBAIO_Format origformat = format; 3208 // adjust format to match that of the Fast Mixer 3209 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3210 format.mFormat = fastMixerFormat; 3211 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3212 3213 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3214 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3215 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3216 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3217 const NBAIO_Format offers[1] = {format}; 3218 size_t numCounterOffers = 0; 3219 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3220 ALOG_ASSERT(index == 0); 3221 monoPipe->setAvgFrames((mScreenState & 1) ? 3222 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3223 mPipeSink = monoPipe; 3224 3225#ifdef TEE_SINK 3226 if (mTeeSinkOutputEnabled) { 3227 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3228 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3229 const NBAIO_Format offers2[1] = {origformat}; 3230 numCounterOffers = 0; 3231 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3232 ALOG_ASSERT(index == 0); 3233 mTeeSink = teeSink; 3234 PipeReader *teeSource = new PipeReader(*teeSink); 3235 numCounterOffers = 0; 3236 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3237 ALOG_ASSERT(index == 0); 3238 mTeeSource = teeSource; 3239 } 3240#endif 3241 3242 // create fast mixer and configure it initially with just one fast track for our submix 3243 mFastMixer = new FastMixer(); 3244 FastMixerStateQueue *sq = mFastMixer->sq(); 3245#ifdef STATE_QUEUE_DUMP 3246 sq->setObserverDump(&mStateQueueObserverDump); 3247 sq->setMutatorDump(&mStateQueueMutatorDump); 3248#endif 3249 FastMixerState *state = sq->begin(); 3250 FastTrack *fastTrack = &state->mFastTracks[0]; 3251 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3252 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3253 fastTrack->mVolumeProvider = NULL; 3254 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3255 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3256 fastTrack->mGeneration++; 3257 state->mFastTracksGen++; 3258 state->mTrackMask = 1; 3259 // fast mixer will use the HAL output sink 3260 state->mOutputSink = mOutputSink.get(); 3261 state->mOutputSinkGen++; 3262 state->mFrameCount = mFrameCount; 3263 state->mCommand = FastMixerState::COLD_IDLE; 3264 // already done in constructor initialization list 3265 //mFastMixerFutex = 0; 3266 state->mColdFutexAddr = &mFastMixerFutex; 3267 state->mColdGen++; 3268 state->mDumpState = &mFastMixerDumpState; 3269#ifdef TEE_SINK 3270 state->mTeeSink = mTeeSink.get(); 3271#endif 3272 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3273 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3274 sq->end(); 3275 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3276 3277 // start the fast mixer 3278 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3279 pid_t tid = mFastMixer->getTid(); 3280 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3281 3282#ifdef AUDIO_WATCHDOG 3283 // create and start the watchdog 3284 mAudioWatchdog = new AudioWatchdog(); 3285 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3286 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3287 tid = mAudioWatchdog->getTid(); 3288 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3289#endif 3290 3291 } 3292 3293 switch (kUseFastMixer) { 3294 case FastMixer_Never: 3295 case FastMixer_Dynamic: 3296 mNormalSink = mOutputSink; 3297 break; 3298 case FastMixer_Always: 3299 mNormalSink = mPipeSink; 3300 break; 3301 case FastMixer_Static: 3302 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3303 break; 3304 } 3305} 3306 3307AudioFlinger::MixerThread::~MixerThread() 3308{ 3309 if (mFastMixer != 0) { 3310 FastMixerStateQueue *sq = mFastMixer->sq(); 3311 FastMixerState *state = sq->begin(); 3312 if (state->mCommand == FastMixerState::COLD_IDLE) { 3313 int32_t old = android_atomic_inc(&mFastMixerFutex); 3314 if (old == -1) { 3315 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3316 } 3317 } 3318 state->mCommand = FastMixerState::EXIT; 3319 sq->end(); 3320 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3321 mFastMixer->join(); 3322 // Though the fast mixer thread has exited, it's state queue is still valid. 3323 // We'll use that extract the final state which contains one remaining fast track 3324 // corresponding to our sub-mix. 3325 state = sq->begin(); 3326 ALOG_ASSERT(state->mTrackMask == 1); 3327 FastTrack *fastTrack = &state->mFastTracks[0]; 3328 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3329 delete fastTrack->mBufferProvider; 3330 sq->end(false /*didModify*/); 3331 mFastMixer.clear(); 3332#ifdef AUDIO_WATCHDOG 3333 if (mAudioWatchdog != 0) { 3334 mAudioWatchdog->requestExit(); 3335 mAudioWatchdog->requestExitAndWait(); 3336 mAudioWatchdog.clear(); 3337 } 3338#endif 3339 } 3340 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3341 delete mAudioMixer; 3342} 3343 3344 3345uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3346{ 3347 if (mFastMixer != 0) { 3348 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3349 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3350 } 3351 return latency; 3352} 3353 3354 3355void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3356{ 3357 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3358} 3359 3360ssize_t AudioFlinger::MixerThread::threadLoop_write() 3361{ 3362 // FIXME we should only do one push per cycle; confirm this is true 3363 // Start the fast mixer if it's not already running 3364 if (mFastMixer != 0) { 3365 FastMixerStateQueue *sq = mFastMixer->sq(); 3366 FastMixerState *state = sq->begin(); 3367 if (state->mCommand != FastMixerState::MIX_WRITE && 3368 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3369 if (state->mCommand == FastMixerState::COLD_IDLE) { 3370 int32_t old = android_atomic_inc(&mFastMixerFutex); 3371 if (old == -1) { 3372 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3373 } 3374#ifdef AUDIO_WATCHDOG 3375 if (mAudioWatchdog != 0) { 3376 mAudioWatchdog->resume(); 3377 } 3378#endif 3379 } 3380 state->mCommand = FastMixerState::MIX_WRITE; 3381#ifdef FAST_THREAD_STATISTICS 3382 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3383 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3384#endif 3385 sq->end(); 3386 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3387 if (kUseFastMixer == FastMixer_Dynamic) { 3388 mNormalSink = mPipeSink; 3389 } 3390 } else { 3391 sq->end(false /*didModify*/); 3392 } 3393 } 3394 return PlaybackThread::threadLoop_write(); 3395} 3396 3397void AudioFlinger::MixerThread::threadLoop_standby() 3398{ 3399 // Idle the fast mixer if it's currently running 3400 if (mFastMixer != 0) { 3401 FastMixerStateQueue *sq = mFastMixer->sq(); 3402 FastMixerState *state = sq->begin(); 3403 if (!(state->mCommand & FastMixerState::IDLE)) { 3404 state->mCommand = FastMixerState::COLD_IDLE; 3405 state->mColdFutexAddr = &mFastMixerFutex; 3406 state->mColdGen++; 3407 mFastMixerFutex = 0; 3408 sq->end(); 3409 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3410 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3411 if (kUseFastMixer == FastMixer_Dynamic) { 3412 mNormalSink = mOutputSink; 3413 } 3414#ifdef AUDIO_WATCHDOG 3415 if (mAudioWatchdog != 0) { 3416 mAudioWatchdog->pause(); 3417 } 3418#endif 3419 } else { 3420 sq->end(false /*didModify*/); 3421 } 3422 } 3423 PlaybackThread::threadLoop_standby(); 3424} 3425 3426bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3427{ 3428 return false; 3429} 3430 3431bool AudioFlinger::PlaybackThread::shouldStandby_l() 3432{ 3433 return !mStandby; 3434} 3435 3436bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3437{ 3438 Mutex::Autolock _l(mLock); 3439 return waitingAsyncCallback_l(); 3440} 3441 3442// shared by MIXER and DIRECT, overridden by DUPLICATING 3443void AudioFlinger::PlaybackThread::threadLoop_standby() 3444{ 3445 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3446 mOutput->standby(); 3447 if (mUseAsyncWrite != 0) { 3448 // discard any pending drain or write ack by incrementing sequence 3449 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3450 mDrainSequence = (mDrainSequence + 2) & ~1; 3451 ALOG_ASSERT(mCallbackThread != 0); 3452 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3453 mCallbackThread->setDraining(mDrainSequence); 3454 } 3455 mHwPaused = false; 3456} 3457 3458void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3459{ 3460 ALOGV("signal playback thread"); 3461 broadcast_l(); 3462} 3463 3464void AudioFlinger::MixerThread::threadLoop_mix() 3465{ 3466 // obtain the presentation timestamp of the next output buffer 3467 int64_t pts; 3468 status_t status = INVALID_OPERATION; 3469 3470 if (mNormalSink != 0) { 3471 status = mNormalSink->getNextWriteTimestamp(&pts); 3472 } else { 3473 status = mOutputSink->getNextWriteTimestamp(&pts); 3474 } 3475 3476 if (status != NO_ERROR) { 3477 pts = AudioBufferProvider::kInvalidPTS; 3478 } 3479 3480 // mix buffers... 3481 mAudioMixer->process(pts); 3482 mCurrentWriteLength = mSinkBufferSize; 3483 // increase sleep time progressively when application underrun condition clears. 3484 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3485 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3486 // such that we would underrun the audio HAL. 3487 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3488 sleepTimeShift--; 3489 } 3490 sleepTime = 0; 3491 standbyTime = systemTime() + standbyDelay; 3492 //TODO: delay standby when effects have a tail 3493 3494} 3495 3496void AudioFlinger::MixerThread::threadLoop_sleepTime() 3497{ 3498 // If no tracks are ready, sleep once for the duration of an output 3499 // buffer size, then write 0s to the output 3500 if (sleepTime == 0) { 3501 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3502 sleepTime = activeSleepTime >> sleepTimeShift; 3503 if (sleepTime < kMinThreadSleepTimeUs) { 3504 sleepTime = kMinThreadSleepTimeUs; 3505 } 3506 // reduce sleep time in case of consecutive application underruns to avoid 3507 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3508 // duration we would end up writing less data than needed by the audio HAL if 3509 // the condition persists. 3510 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3511 sleepTimeShift++; 3512 } 3513 } else { 3514 sleepTime = idleSleepTime; 3515 } 3516 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3517 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3518 // before effects processing or output. 3519 if (mMixerBufferValid) { 3520 memset(mMixerBuffer, 0, mMixerBufferSize); 3521 } else { 3522 memset(mSinkBuffer, 0, mSinkBufferSize); 3523 } 3524 sleepTime = 0; 3525 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3526 "anticipated start"); 3527 } 3528 // TODO add standby time extension fct of effect tail 3529} 3530 3531// prepareTracks_l() must be called with ThreadBase::mLock held 3532AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3533 Vector< sp<Track> > *tracksToRemove) 3534{ 3535 3536 mixer_state mixerStatus = MIXER_IDLE; 3537 // find out which tracks need to be processed 3538 size_t count = mActiveTracks.size(); 3539 size_t mixedTracks = 0; 3540 size_t tracksWithEffect = 0; 3541 // counts only _active_ fast tracks 3542 size_t fastTracks = 0; 3543 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3544 3545 float masterVolume = mMasterVolume; 3546 bool masterMute = mMasterMute; 3547 3548 if (masterMute) { 3549 masterVolume = 0; 3550 } 3551 // Delegate master volume control to effect in output mix effect chain if needed 3552 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3553 if (chain != 0) { 3554 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3555 chain->setVolume_l(&v, &v); 3556 masterVolume = (float)((v + (1 << 23)) >> 24); 3557 chain.clear(); 3558 } 3559 3560 // prepare a new state to push 3561 FastMixerStateQueue *sq = NULL; 3562 FastMixerState *state = NULL; 3563 bool didModify = false; 3564 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3565 if (mFastMixer != 0) { 3566 sq = mFastMixer->sq(); 3567 state = sq->begin(); 3568 } 3569 3570 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3571 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3572 3573 for (size_t i=0 ; i<count ; i++) { 3574 const sp<Track> t = mActiveTracks[i].promote(); 3575 if (t == 0) { 3576 continue; 3577 } 3578 3579 // this const just means the local variable doesn't change 3580 Track* const track = t.get(); 3581 3582 // process fast tracks 3583 if (track->isFastTrack()) { 3584 3585 // It's theoretically possible (though unlikely) for a fast track to be created 3586 // and then removed within the same normal mix cycle. This is not a problem, as 3587 // the track never becomes active so it's fast mixer slot is never touched. 3588 // The converse, of removing an (active) track and then creating a new track 3589 // at the identical fast mixer slot within the same normal mix cycle, 3590 // is impossible because the slot isn't marked available until the end of each cycle. 3591 int j = track->mFastIndex; 3592 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3593 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3594 FastTrack *fastTrack = &state->mFastTracks[j]; 3595 3596 // Determine whether the track is currently in underrun condition, 3597 // and whether it had a recent underrun. 3598 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3599 FastTrackUnderruns underruns = ftDump->mUnderruns; 3600 uint32_t recentFull = (underruns.mBitFields.mFull - 3601 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3602 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3603 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3604 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3605 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3606 uint32_t recentUnderruns = recentPartial + recentEmpty; 3607 track->mObservedUnderruns = underruns; 3608 // don't count underruns that occur while stopping or pausing 3609 // or stopped which can occur when flush() is called while active 3610 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3611 recentUnderruns > 0) { 3612 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3613 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3614 } 3615 3616 // This is similar to the state machine for normal tracks, 3617 // with a few modifications for fast tracks. 3618 bool isActive = true; 3619 switch (track->mState) { 3620 case TrackBase::STOPPING_1: 3621 // track stays active in STOPPING_1 state until first underrun 3622 if (recentUnderruns > 0 || track->isTerminated()) { 3623 track->mState = TrackBase::STOPPING_2; 3624 } 3625 break; 3626 case TrackBase::PAUSING: 3627 // ramp down is not yet implemented 3628 track->setPaused(); 3629 break; 3630 case TrackBase::RESUMING: 3631 // ramp up is not yet implemented 3632 track->mState = TrackBase::ACTIVE; 3633 break; 3634 case TrackBase::ACTIVE: 3635 if (recentFull > 0 || recentPartial > 0) { 3636 // track has provided at least some frames recently: reset retry count 3637 track->mRetryCount = kMaxTrackRetries; 3638 } 3639 if (recentUnderruns == 0) { 3640 // no recent underruns: stay active 3641 break; 3642 } 3643 // there has recently been an underrun of some kind 3644 if (track->sharedBuffer() == 0) { 3645 // were any of the recent underruns "empty" (no frames available)? 3646 if (recentEmpty == 0) { 3647 // no, then ignore the partial underruns as they are allowed indefinitely 3648 break; 3649 } 3650 // there has recently been an "empty" underrun: decrement the retry counter 3651 if (--(track->mRetryCount) > 0) { 3652 break; 3653 } 3654 // indicate to client process that the track was disabled because of underrun; 3655 // it will then automatically call start() when data is available 3656 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3657 // remove from active list, but state remains ACTIVE [confusing but true] 3658 isActive = false; 3659 break; 3660 } 3661 // fall through 3662 case TrackBase::STOPPING_2: 3663 case TrackBase::PAUSED: 3664 case TrackBase::STOPPED: 3665 case TrackBase::FLUSHED: // flush() while active 3666 // Check for presentation complete if track is inactive 3667 // We have consumed all the buffers of this track. 3668 // This would be incomplete if we auto-paused on underrun 3669 { 3670 size_t audioHALFrames = 3671 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3672 size_t framesWritten = mBytesWritten / mFrameSize; 3673 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3674 // track stays in active list until presentation is complete 3675 break; 3676 } 3677 } 3678 if (track->isStopping_2()) { 3679 track->mState = TrackBase::STOPPED; 3680 } 3681 if (track->isStopped()) { 3682 // Can't reset directly, as fast mixer is still polling this track 3683 // track->reset(); 3684 // So instead mark this track as needing to be reset after push with ack 3685 resetMask |= 1 << i; 3686 } 3687 isActive = false; 3688 break; 3689 case TrackBase::IDLE: 3690 default: 3691 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3692 } 3693 3694 if (isActive) { 3695 // was it previously inactive? 3696 if (!(state->mTrackMask & (1 << j))) { 3697 ExtendedAudioBufferProvider *eabp = track; 3698 VolumeProvider *vp = track; 3699 fastTrack->mBufferProvider = eabp; 3700 fastTrack->mVolumeProvider = vp; 3701 fastTrack->mChannelMask = track->mChannelMask; 3702 fastTrack->mFormat = track->mFormat; 3703 fastTrack->mGeneration++; 3704 state->mTrackMask |= 1 << j; 3705 didModify = true; 3706 // no acknowledgement required for newly active tracks 3707 } 3708 // cache the combined master volume and stream type volume for fast mixer; this 3709 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3710 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3711 ++fastTracks; 3712 } else { 3713 // was it previously active? 3714 if (state->mTrackMask & (1 << j)) { 3715 fastTrack->mBufferProvider = NULL; 3716 fastTrack->mGeneration++; 3717 state->mTrackMask &= ~(1 << j); 3718 didModify = true; 3719 // If any fast tracks were removed, we must wait for acknowledgement 3720 // because we're about to decrement the last sp<> on those tracks. 3721 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3722 } else { 3723 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3724 } 3725 tracksToRemove->add(track); 3726 // Avoids a misleading display in dumpsys 3727 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3728 } 3729 continue; 3730 } 3731 3732 { // local variable scope to avoid goto warning 3733 3734 audio_track_cblk_t* cblk = track->cblk(); 3735 3736 // The first time a track is added we wait 3737 // for all its buffers to be filled before processing it 3738 int name = track->name(); 3739 // make sure that we have enough frames to mix one full buffer. 3740 // enforce this condition only once to enable draining the buffer in case the client 3741 // app does not call stop() and relies on underrun to stop: 3742 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3743 // during last round 3744 size_t desiredFrames; 3745 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3746 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3747 3748 desiredFrames = sourceFramesNeededWithTimestretch( 3749 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3750 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3751 // add frames already consumed but not yet released by the resampler 3752 // because mAudioTrackServerProxy->framesReady() will include these frames 3753 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3754 3755 uint32_t minFrames = 1; 3756 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3757 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3758 minFrames = desiredFrames; 3759 } 3760 3761 size_t framesReady = track->framesReady(); 3762 if (ATRACE_ENABLED()) { 3763 // I wish we had formatted trace names 3764 char traceName[16]; 3765 strcpy(traceName, "nRdy"); 3766 int name = track->name(); 3767 if (AudioMixer::TRACK0 <= name && 3768 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3769 name -= AudioMixer::TRACK0; 3770 traceName[4] = (name / 10) + '0'; 3771 traceName[5] = (name % 10) + '0'; 3772 } else { 3773 traceName[4] = '?'; 3774 traceName[5] = '?'; 3775 } 3776 traceName[6] = '\0'; 3777 ATRACE_INT(traceName, framesReady); 3778 } 3779 if ((framesReady >= minFrames) && track->isReady() && 3780 !track->isPaused() && !track->isTerminated()) 3781 { 3782 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3783 3784 mixedTracks++; 3785 3786 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3787 // there is an effect chain connected to the track 3788 chain.clear(); 3789 if (track->mainBuffer() != mSinkBuffer && 3790 track->mainBuffer() != mMixerBuffer) { 3791 if (mEffectBufferEnabled) { 3792 mEffectBufferValid = true; // Later can set directly. 3793 } 3794 chain = getEffectChain_l(track->sessionId()); 3795 // Delegate volume control to effect in track effect chain if needed 3796 if (chain != 0) { 3797 tracksWithEffect++; 3798 } else { 3799 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3800 "session %d", 3801 name, track->sessionId()); 3802 } 3803 } 3804 3805 3806 int param = AudioMixer::VOLUME; 3807 if (track->mFillingUpStatus == Track::FS_FILLED) { 3808 // no ramp for the first volume setting 3809 track->mFillingUpStatus = Track::FS_ACTIVE; 3810 if (track->mState == TrackBase::RESUMING) { 3811 track->mState = TrackBase::ACTIVE; 3812 param = AudioMixer::RAMP_VOLUME; 3813 } 3814 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3815 // FIXME should not make a decision based on mServer 3816 } else if (cblk->mServer != 0) { 3817 // If the track is stopped before the first frame was mixed, 3818 // do not apply ramp 3819 param = AudioMixer::RAMP_VOLUME; 3820 } 3821 3822 // compute volume for this track 3823 uint32_t vl, vr; // in U8.24 integer format 3824 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3825 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3826 vl = vr = 0; 3827 vlf = vrf = vaf = 0.; 3828 if (track->isPausing()) { 3829 track->setPaused(); 3830 } 3831 } else { 3832 3833 // read original volumes with volume control 3834 float typeVolume = mStreamTypes[track->streamType()].volume; 3835 float v = masterVolume * typeVolume; 3836 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3837 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3838 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3839 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3840 // track volumes come from shared memory, so can't be trusted and must be clamped 3841 if (vlf > GAIN_FLOAT_UNITY) { 3842 ALOGV("Track left volume out of range: %.3g", vlf); 3843 vlf = GAIN_FLOAT_UNITY; 3844 } 3845 if (vrf > GAIN_FLOAT_UNITY) { 3846 ALOGV("Track right volume out of range: %.3g", vrf); 3847 vrf = GAIN_FLOAT_UNITY; 3848 } 3849 // now apply the master volume and stream type volume 3850 vlf *= v; 3851 vrf *= v; 3852 // assuming master volume and stream type volume each go up to 1.0, 3853 // then derive vl and vr as U8.24 versions for the effect chain 3854 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3855 vl = (uint32_t) (scaleto8_24 * vlf); 3856 vr = (uint32_t) (scaleto8_24 * vrf); 3857 // vl and vr are now in U8.24 format 3858 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3859 // send level comes from shared memory and so may be corrupt 3860 if (sendLevel > MAX_GAIN_INT) { 3861 ALOGV("Track send level out of range: %04X", sendLevel); 3862 sendLevel = MAX_GAIN_INT; 3863 } 3864 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3865 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3866 } 3867 3868 // Delegate volume control to effect in track effect chain if needed 3869 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3870 // Do not ramp volume if volume is controlled by effect 3871 param = AudioMixer::VOLUME; 3872 // Update remaining floating point volume levels 3873 vlf = (float)vl / (1 << 24); 3874 vrf = (float)vr / (1 << 24); 3875 track->mHasVolumeController = true; 3876 } else { 3877 // force no volume ramp when volume controller was just disabled or removed 3878 // from effect chain to avoid volume spike 3879 if (track->mHasVolumeController) { 3880 param = AudioMixer::VOLUME; 3881 } 3882 track->mHasVolumeController = false; 3883 } 3884 3885 // XXX: these things DON'T need to be done each time 3886 mAudioMixer->setBufferProvider(name, track); 3887 mAudioMixer->enable(name); 3888 3889 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3890 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3891 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3892 mAudioMixer->setParameter( 3893 name, 3894 AudioMixer::TRACK, 3895 AudioMixer::FORMAT, (void *)track->format()); 3896 mAudioMixer->setParameter( 3897 name, 3898 AudioMixer::TRACK, 3899 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3900 mAudioMixer->setParameter( 3901 name, 3902 AudioMixer::TRACK, 3903 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3904 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3905 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3906 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3907 if (reqSampleRate == 0) { 3908 reqSampleRate = mSampleRate; 3909 } else if (reqSampleRate > maxSampleRate) { 3910 reqSampleRate = maxSampleRate; 3911 } 3912 mAudioMixer->setParameter( 3913 name, 3914 AudioMixer::RESAMPLE, 3915 AudioMixer::SAMPLE_RATE, 3916 (void *)(uintptr_t)reqSampleRate); 3917 3918 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3919 mAudioMixer->setParameter( 3920 name, 3921 AudioMixer::TIMESTRETCH, 3922 AudioMixer::PLAYBACK_RATE, 3923 &playbackRate); 3924 3925 /* 3926 * Select the appropriate output buffer for the track. 3927 * 3928 * Tracks with effects go into their own effects chain buffer 3929 * and from there into either mEffectBuffer or mSinkBuffer. 3930 * 3931 * Other tracks can use mMixerBuffer for higher precision 3932 * channel accumulation. If this buffer is enabled 3933 * (mMixerBufferEnabled true), then selected tracks will accumulate 3934 * into it. 3935 * 3936 */ 3937 if (mMixerBufferEnabled 3938 && (track->mainBuffer() == mSinkBuffer 3939 || track->mainBuffer() == mMixerBuffer)) { 3940 mAudioMixer->setParameter( 3941 name, 3942 AudioMixer::TRACK, 3943 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3944 mAudioMixer->setParameter( 3945 name, 3946 AudioMixer::TRACK, 3947 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3948 // TODO: override track->mainBuffer()? 3949 mMixerBufferValid = true; 3950 } else { 3951 mAudioMixer->setParameter( 3952 name, 3953 AudioMixer::TRACK, 3954 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3955 mAudioMixer->setParameter( 3956 name, 3957 AudioMixer::TRACK, 3958 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3959 } 3960 mAudioMixer->setParameter( 3961 name, 3962 AudioMixer::TRACK, 3963 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3964 3965 // reset retry count 3966 track->mRetryCount = kMaxTrackRetries; 3967 3968 // If one track is ready, set the mixer ready if: 3969 // - the mixer was not ready during previous round OR 3970 // - no other track is not ready 3971 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3972 mixerStatus != MIXER_TRACKS_ENABLED) { 3973 mixerStatus = MIXER_TRACKS_READY; 3974 } 3975 } else { 3976 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3977 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3978 } 3979 // clear effect chain input buffer if an active track underruns to avoid sending 3980 // previous audio buffer again to effects 3981 chain = getEffectChain_l(track->sessionId()); 3982 if (chain != 0) { 3983 chain->clearInputBuffer(); 3984 } 3985 3986 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3987 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3988 track->isStopped() || track->isPaused()) { 3989 // We have consumed all the buffers of this track. 3990 // Remove it from the list of active tracks. 3991 // TODO: use actual buffer filling status instead of latency when available from 3992 // audio HAL 3993 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3994 size_t framesWritten = mBytesWritten / mFrameSize; 3995 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3996 if (track->isStopped()) { 3997 track->reset(); 3998 } 3999 tracksToRemove->add(track); 4000 } 4001 } else { 4002 // No buffers for this track. Give it a few chances to 4003 // fill a buffer, then remove it from active list. 4004 if (--(track->mRetryCount) <= 0) { 4005 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4006 tracksToRemove->add(track); 4007 // indicate to client process that the track was disabled because of underrun; 4008 // it will then automatically call start() when data is available 4009 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4010 // If one track is not ready, mark the mixer also not ready if: 4011 // - the mixer was ready during previous round OR 4012 // - no other track is ready 4013 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4014 mixerStatus != MIXER_TRACKS_READY) { 4015 mixerStatus = MIXER_TRACKS_ENABLED; 4016 } 4017 } 4018 mAudioMixer->disable(name); 4019 } 4020 4021 } // local variable scope to avoid goto warning 4022track_is_ready: ; 4023 4024 } 4025 4026 // Push the new FastMixer state if necessary 4027 bool pauseAudioWatchdog = false; 4028 if (didModify) { 4029 state->mFastTracksGen++; 4030 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4031 if (kUseFastMixer == FastMixer_Dynamic && 4032 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4033 state->mCommand = FastMixerState::COLD_IDLE; 4034 state->mColdFutexAddr = &mFastMixerFutex; 4035 state->mColdGen++; 4036 mFastMixerFutex = 0; 4037 if (kUseFastMixer == FastMixer_Dynamic) { 4038 mNormalSink = mOutputSink; 4039 } 4040 // If we go into cold idle, need to wait for acknowledgement 4041 // so that fast mixer stops doing I/O. 4042 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4043 pauseAudioWatchdog = true; 4044 } 4045 } 4046 if (sq != NULL) { 4047 sq->end(didModify); 4048 sq->push(block); 4049 } 4050#ifdef AUDIO_WATCHDOG 4051 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4052 mAudioWatchdog->pause(); 4053 } 4054#endif 4055 4056 // Now perform the deferred reset on fast tracks that have stopped 4057 while (resetMask != 0) { 4058 size_t i = __builtin_ctz(resetMask); 4059 ALOG_ASSERT(i < count); 4060 resetMask &= ~(1 << i); 4061 sp<Track> t = mActiveTracks[i].promote(); 4062 if (t == 0) { 4063 continue; 4064 } 4065 Track* track = t.get(); 4066 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4067 track->reset(); 4068 } 4069 4070 // remove all the tracks that need to be... 4071 removeTracks_l(*tracksToRemove); 4072 4073 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4074 mEffectBufferValid = true; 4075 } 4076 4077 if (mEffectBufferValid) { 4078 // as long as there are effects we should clear the effects buffer, to avoid 4079 // passing a non-clean buffer to the effect chain 4080 memset(mEffectBuffer, 0, mEffectBufferSize); 4081 } 4082 // sink or mix buffer must be cleared if all tracks are connected to an 4083 // effect chain as in this case the mixer will not write to the sink or mix buffer 4084 // and track effects will accumulate into it 4085 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4086 (mixedTracks == 0 && fastTracks > 0))) { 4087 // FIXME as a performance optimization, should remember previous zero status 4088 if (mMixerBufferValid) { 4089 memset(mMixerBuffer, 0, mMixerBufferSize); 4090 // TODO: In testing, mSinkBuffer below need not be cleared because 4091 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4092 // after mixing. 4093 // 4094 // To enforce this guarantee: 4095 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4096 // (mixedTracks == 0 && fastTracks > 0)) 4097 // must imply MIXER_TRACKS_READY. 4098 // Later, we may clear buffers regardless, and skip much of this logic. 4099 } 4100 // FIXME as a performance optimization, should remember previous zero status 4101 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4102 } 4103 4104 // if any fast tracks, then status is ready 4105 mMixerStatusIgnoringFastTracks = mixerStatus; 4106 if (fastTracks > 0) { 4107 mixerStatus = MIXER_TRACKS_READY; 4108 } 4109 return mixerStatus; 4110} 4111 4112// getTrackName_l() must be called with ThreadBase::mLock held 4113int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4114 audio_format_t format, int sessionId) 4115{ 4116 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4117} 4118 4119// deleteTrackName_l() must be called with ThreadBase::mLock held 4120void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4121{ 4122 ALOGV("remove track (%d) and delete from mixer", name); 4123 mAudioMixer->deleteTrackName(name); 4124} 4125 4126// checkForNewParameter_l() must be called with ThreadBase::mLock held 4127bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4128 status_t& status) 4129{ 4130 bool reconfig = false; 4131 4132 status = NO_ERROR; 4133 4134 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4135 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4136 if (mFastMixer != 0) { 4137 FastMixerStateQueue *sq = mFastMixer->sq(); 4138 FastMixerState *state = sq->begin(); 4139 if (!(state->mCommand & FastMixerState::IDLE)) { 4140 previousCommand = state->mCommand; 4141 state->mCommand = FastMixerState::HOT_IDLE; 4142 sq->end(); 4143 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4144 } else { 4145 sq->end(false /*didModify*/); 4146 } 4147 } 4148 4149 AudioParameter param = AudioParameter(keyValuePair); 4150 int value; 4151 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4152 reconfig = true; 4153 } 4154 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4155 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4156 status = BAD_VALUE; 4157 } else { 4158 // no need to save value, since it's constant 4159 reconfig = true; 4160 } 4161 } 4162 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4163 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4164 status = BAD_VALUE; 4165 } else { 4166 // no need to save value, since it's constant 4167 reconfig = true; 4168 } 4169 } 4170 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4171 // do not accept frame count changes if tracks are open as the track buffer 4172 // size depends on frame count and correct behavior would not be guaranteed 4173 // if frame count is changed after track creation 4174 if (!mTracks.isEmpty()) { 4175 status = INVALID_OPERATION; 4176 } else { 4177 reconfig = true; 4178 } 4179 } 4180 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4181#ifdef ADD_BATTERY_DATA 4182 // when changing the audio output device, call addBatteryData to notify 4183 // the change 4184 if (mOutDevice != value) { 4185 uint32_t params = 0; 4186 // check whether speaker is on 4187 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4188 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4189 } 4190 4191 audio_devices_t deviceWithoutSpeaker 4192 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4193 // check if any other device (except speaker) is on 4194 if (value & deviceWithoutSpeaker) { 4195 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4196 } 4197 4198 if (params != 0) { 4199 addBatteryData(params); 4200 } 4201 } 4202#endif 4203 4204 // forward device change to effects that have requested to be 4205 // aware of attached audio device. 4206 if (value != AUDIO_DEVICE_NONE) { 4207 mOutDevice = value; 4208 for (size_t i = 0; i < mEffectChains.size(); i++) { 4209 mEffectChains[i]->setDevice_l(mOutDevice); 4210 } 4211 } 4212 } 4213 4214 if (status == NO_ERROR) { 4215 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4216 keyValuePair.string()); 4217 if (!mStandby && status == INVALID_OPERATION) { 4218 mOutput->standby(); 4219 mStandby = true; 4220 mBytesWritten = 0; 4221 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4222 keyValuePair.string()); 4223 } 4224 if (status == NO_ERROR && reconfig) { 4225 readOutputParameters_l(); 4226 delete mAudioMixer; 4227 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4228 for (size_t i = 0; i < mTracks.size() ; i++) { 4229 int name = getTrackName_l(mTracks[i]->mChannelMask, 4230 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4231 if (name < 0) { 4232 break; 4233 } 4234 mTracks[i]->mName = name; 4235 } 4236 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4237 } 4238 } 4239 4240 if (!(previousCommand & FastMixerState::IDLE)) { 4241 ALOG_ASSERT(mFastMixer != 0); 4242 FastMixerStateQueue *sq = mFastMixer->sq(); 4243 FastMixerState *state = sq->begin(); 4244 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4245 state->mCommand = previousCommand; 4246 sq->end(); 4247 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4248 } 4249 4250 return reconfig; 4251} 4252 4253 4254void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4255{ 4256 const size_t SIZE = 256; 4257 char buffer[SIZE]; 4258 String8 result; 4259 4260 PlaybackThread::dumpInternals(fd, args); 4261 4262 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4263 4264 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4265 const FastMixerDumpState copy(mFastMixerDumpState); 4266 copy.dump(fd); 4267 4268#ifdef STATE_QUEUE_DUMP 4269 // Similar for state queue 4270 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4271 observerCopy.dump(fd); 4272 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4273 mutatorCopy.dump(fd); 4274#endif 4275 4276#ifdef TEE_SINK 4277 // Write the tee output to a .wav file 4278 dumpTee(fd, mTeeSource, mId); 4279#endif 4280 4281#ifdef AUDIO_WATCHDOG 4282 if (mAudioWatchdog != 0) { 4283 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4284 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4285 wdCopy.dump(fd); 4286 } 4287#endif 4288} 4289 4290uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4291{ 4292 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4293} 4294 4295uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4296{ 4297 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4298} 4299 4300void AudioFlinger::MixerThread::cacheParameters_l() 4301{ 4302 PlaybackThread::cacheParameters_l(); 4303 4304 // FIXME: Relaxed timing because of a certain device that can't meet latency 4305 // Should be reduced to 2x after the vendor fixes the driver issue 4306 // increase threshold again due to low power audio mode. The way this warning 4307 // threshold is calculated and its usefulness should be reconsidered anyway. 4308 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4309} 4310 4311// ---------------------------------------------------------------------------- 4312 4313AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4314 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4315 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4316 // mLeftVolFloat, mRightVolFloat 4317{ 4318} 4319 4320AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4321 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4322 ThreadBase::type_t type, bool systemReady) 4323 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4324 // mLeftVolFloat, mRightVolFloat 4325{ 4326} 4327 4328AudioFlinger::DirectOutputThread::~DirectOutputThread() 4329{ 4330} 4331 4332void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4333{ 4334 audio_track_cblk_t* cblk = track->cblk(); 4335 float left, right; 4336 4337 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4338 left = right = 0; 4339 } else { 4340 float typeVolume = mStreamTypes[track->streamType()].volume; 4341 float v = mMasterVolume * typeVolume; 4342 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4343 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4344 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4345 if (left > GAIN_FLOAT_UNITY) { 4346 left = GAIN_FLOAT_UNITY; 4347 } 4348 left *= v; 4349 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4350 if (right > GAIN_FLOAT_UNITY) { 4351 right = GAIN_FLOAT_UNITY; 4352 } 4353 right *= v; 4354 } 4355 4356 if (lastTrack) { 4357 if (left != mLeftVolFloat || right != mRightVolFloat) { 4358 mLeftVolFloat = left; 4359 mRightVolFloat = right; 4360 4361 // Convert volumes from float to 8.24 4362 uint32_t vl = (uint32_t)(left * (1 << 24)); 4363 uint32_t vr = (uint32_t)(right * (1 << 24)); 4364 4365 // Delegate volume control to effect in track effect chain if needed 4366 // only one effect chain can be present on DirectOutputThread, so if 4367 // there is one, the track is connected to it 4368 if (!mEffectChains.isEmpty()) { 4369 mEffectChains[0]->setVolume_l(&vl, &vr); 4370 left = (float)vl / (1 << 24); 4371 right = (float)vr / (1 << 24); 4372 } 4373 if (mOutput->stream->set_volume) { 4374 mOutput->stream->set_volume(mOutput->stream, left, right); 4375 } 4376 } 4377 } 4378} 4379 4380 4381AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4382 Vector< sp<Track> > *tracksToRemove 4383) 4384{ 4385 size_t count = mActiveTracks.size(); 4386 mixer_state mixerStatus = MIXER_IDLE; 4387 bool doHwPause = false; 4388 bool doHwResume = false; 4389 bool flushPending = false; 4390 4391 // find out which tracks need to be processed 4392 for (size_t i = 0; i < count; i++) { 4393 sp<Track> t = mActiveTracks[i].promote(); 4394 // The track died recently 4395 if (t == 0) { 4396 continue; 4397 } 4398 4399 Track* const track = t.get(); 4400 audio_track_cblk_t* cblk = track->cblk(); 4401 // Only consider last track started for volume and mixer state control. 4402 // In theory an older track could underrun and restart after the new one starts 4403 // but as we only care about the transition phase between two tracks on a 4404 // direct output, it is not a problem to ignore the underrun case. 4405 sp<Track> l = mLatestActiveTrack.promote(); 4406 bool last = l.get() == track; 4407 4408 if (track->isPausing()) { 4409 track->setPaused(); 4410 if (mHwSupportsPause && last && !mHwPaused) { 4411 doHwPause = true; 4412 mHwPaused = true; 4413 } 4414 tracksToRemove->add(track); 4415 } else if (track->isFlushPending()) { 4416 track->flushAck(); 4417 if (last) { 4418 flushPending = true; 4419 } 4420 } else if (track->isResumePending()) { 4421 track->resumeAck(); 4422 if (last && mHwPaused) { 4423 doHwResume = true; 4424 mHwPaused = false; 4425 } 4426 } 4427 4428 // The first time a track is added we wait 4429 // for all its buffers to be filled before processing it. 4430 // Allow draining the buffer in case the client 4431 // app does not call stop() and relies on underrun to stop: 4432 // hence the test on (track->mRetryCount > 1). 4433 // If retryCount<=1 then track is about to underrun and be removed. 4434 uint32_t minFrames; 4435 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4436 && (track->mRetryCount > 1)) { 4437 minFrames = mNormalFrameCount; 4438 } else { 4439 minFrames = 1; 4440 } 4441 4442 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4443 !track->isStopping_2() && !track->isStopped()) 4444 { 4445 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4446 4447 if (track->mFillingUpStatus == Track::FS_FILLED) { 4448 track->mFillingUpStatus = Track::FS_ACTIVE; 4449 // make sure processVolume_l() will apply new volume even if 0 4450 mLeftVolFloat = mRightVolFloat = -1.0; 4451 if (!mHwSupportsPause) { 4452 track->resumeAck(); 4453 } 4454 } 4455 4456 // compute volume for this track 4457 processVolume_l(track, last); 4458 if (last) { 4459 // reset retry count 4460 track->mRetryCount = kMaxTrackRetriesDirect; 4461 mActiveTrack = t; 4462 mixerStatus = MIXER_TRACKS_READY; 4463 if (usesHwAvSync() && mHwPaused) { 4464 doHwResume = true; 4465 mHwPaused = false; 4466 } 4467 } 4468 } else { 4469 // clear effect chain input buffer if the last active track started underruns 4470 // to avoid sending previous audio buffer again to effects 4471 if (!mEffectChains.isEmpty() && last) { 4472 mEffectChains[0]->clearInputBuffer(); 4473 } 4474 if (track->isStopping_1()) { 4475 track->mState = TrackBase::STOPPING_2; 4476 if (last && mHwPaused) { 4477 doHwResume = true; 4478 mHwPaused = false; 4479 } 4480 } 4481 if ((track->sharedBuffer() != 0) || track->isStopped() || 4482 track->isStopping_2() || track->isPaused()) { 4483 // We have consumed all the buffers of this track. 4484 // Remove it from the list of active tracks. 4485 size_t audioHALFrames; 4486 if (audio_is_linear_pcm(mFormat)) { 4487 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4488 } else { 4489 audioHALFrames = 0; 4490 } 4491 4492 size_t framesWritten = mBytesWritten / mFrameSize; 4493 if (mStandby || !last || 4494 track->presentationComplete(framesWritten, audioHALFrames)) { 4495 if (track->isStopping_2()) { 4496 track->mState = TrackBase::STOPPED; 4497 } 4498 if (track->isStopped()) { 4499 track->reset(); 4500 } 4501 tracksToRemove->add(track); 4502 } 4503 } else { 4504 // No buffers for this track. Give it a few chances to 4505 // fill a buffer, then remove it from active list. 4506 // Only consider last track started for mixer state control 4507 if (--(track->mRetryCount) <= 0) { 4508 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4509 tracksToRemove->add(track); 4510 // indicate to client process that the track was disabled because of underrun; 4511 // it will then automatically call start() when data is available 4512 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4513 } else if (last) { 4514 mixerStatus = MIXER_TRACKS_ENABLED; 4515 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4516 doHwPause = true; 4517 mHwPaused = true; 4518 } 4519 } 4520 } 4521 } 4522 } 4523 4524 // if an active track did not command a flush, check for pending flush on stopped tracks 4525 if (!flushPending) { 4526 for (size_t i = 0; i < mTracks.size(); i++) { 4527 if (mTracks[i]->isFlushPending()) { 4528 mTracks[i]->flushAck(); 4529 flushPending = true; 4530 } 4531 } 4532 } 4533 4534 // make sure the pause/flush/resume sequence is executed in the right order. 4535 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4536 // before flush and then resume HW. This can happen in case of pause/flush/resume 4537 // if resume is received before pause is executed. 4538 if (mHwSupportsPause && !mStandby && 4539 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4540 mOutput->stream->pause(mOutput->stream); 4541 } 4542 if (flushPending) { 4543 flushHw_l(); 4544 } 4545 if (mHwSupportsPause && !mStandby && doHwResume) { 4546 mOutput->stream->resume(mOutput->stream); 4547 } 4548 // remove all the tracks that need to be... 4549 removeTracks_l(*tracksToRemove); 4550 4551 return mixerStatus; 4552} 4553 4554void AudioFlinger::DirectOutputThread::threadLoop_mix() 4555{ 4556 size_t frameCount = mFrameCount; 4557 int8_t *curBuf = (int8_t *)mSinkBuffer; 4558 // output audio to hardware 4559 while (frameCount) { 4560 AudioBufferProvider::Buffer buffer; 4561 buffer.frameCount = frameCount; 4562 status_t status = mActiveTrack->getNextBuffer(&buffer); 4563 if (status != NO_ERROR || buffer.raw == NULL) { 4564 memset(curBuf, 0, frameCount * mFrameSize); 4565 break; 4566 } 4567 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4568 frameCount -= buffer.frameCount; 4569 curBuf += buffer.frameCount * mFrameSize; 4570 mActiveTrack->releaseBuffer(&buffer); 4571 } 4572 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4573 sleepTime = 0; 4574 standbyTime = systemTime() + standbyDelay; 4575 mActiveTrack.clear(); 4576} 4577 4578void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4579{ 4580 // do not write to HAL when paused 4581 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4582 sleepTime = idleSleepTime; 4583 return; 4584 } 4585 if (sleepTime == 0) { 4586 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4587 sleepTime = activeSleepTime; 4588 } else { 4589 sleepTime = idleSleepTime; 4590 } 4591 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4592 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4593 sleepTime = 0; 4594 } 4595} 4596 4597void AudioFlinger::DirectOutputThread::threadLoop_exit() 4598{ 4599 { 4600 Mutex::Autolock _l(mLock); 4601 bool flushPending = false; 4602 for (size_t i = 0; i < mTracks.size(); i++) { 4603 if (mTracks[i]->isFlushPending()) { 4604 mTracks[i]->flushAck(); 4605 flushPending = true; 4606 } 4607 } 4608 if (flushPending) { 4609 flushHw_l(); 4610 } 4611 } 4612 PlaybackThread::threadLoop_exit(); 4613} 4614 4615// must be called with thread mutex locked 4616bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4617{ 4618 bool trackPaused = false; 4619 bool trackStopped = false; 4620 4621 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4622 // after a timeout and we will enter standby then. 4623 if (mTracks.size() > 0) { 4624 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4625 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4626 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4627 } 4628 4629 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped)); 4630} 4631 4632// getTrackName_l() must be called with ThreadBase::mLock held 4633int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4634 audio_format_t format __unused, int sessionId __unused) 4635{ 4636 return 0; 4637} 4638 4639// deleteTrackName_l() must be called with ThreadBase::mLock held 4640void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4641{ 4642} 4643 4644// checkForNewParameter_l() must be called with ThreadBase::mLock held 4645bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4646 status_t& status) 4647{ 4648 bool reconfig = false; 4649 4650 status = NO_ERROR; 4651 4652 AudioParameter param = AudioParameter(keyValuePair); 4653 int value; 4654 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4655 // forward device change to effects that have requested to be 4656 // aware of attached audio device. 4657 if (value != AUDIO_DEVICE_NONE) { 4658 mOutDevice = value; 4659 for (size_t i = 0; i < mEffectChains.size(); i++) { 4660 mEffectChains[i]->setDevice_l(mOutDevice); 4661 } 4662 } 4663 } 4664 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4665 // do not accept frame count changes if tracks are open as the track buffer 4666 // size depends on frame count and correct behavior would not be garantied 4667 // if frame count is changed after track creation 4668 if (!mTracks.isEmpty()) { 4669 status = INVALID_OPERATION; 4670 } else { 4671 reconfig = true; 4672 } 4673 } 4674 if (status == NO_ERROR) { 4675 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4676 keyValuePair.string()); 4677 if (!mStandby && status == INVALID_OPERATION) { 4678 mOutput->standby(); 4679 mStandby = true; 4680 mBytesWritten = 0; 4681 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4682 keyValuePair.string()); 4683 } 4684 if (status == NO_ERROR && reconfig) { 4685 readOutputParameters_l(); 4686 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4687 } 4688 } 4689 4690 return reconfig; 4691} 4692 4693uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4694{ 4695 uint32_t time; 4696 if (audio_is_linear_pcm(mFormat)) { 4697 time = PlaybackThread::activeSleepTimeUs(); 4698 } else { 4699 time = 10000; 4700 } 4701 return time; 4702} 4703 4704uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4705{ 4706 uint32_t time; 4707 if (audio_is_linear_pcm(mFormat)) { 4708 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4709 } else { 4710 time = 10000; 4711 } 4712 return time; 4713} 4714 4715uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4716{ 4717 uint32_t time; 4718 if (audio_is_linear_pcm(mFormat)) { 4719 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4720 } else { 4721 time = 10000; 4722 } 4723 return time; 4724} 4725 4726void AudioFlinger::DirectOutputThread::cacheParameters_l() 4727{ 4728 PlaybackThread::cacheParameters_l(); 4729 4730 // use shorter standby delay as on normal output to release 4731 // hardware resources as soon as possible 4732 // no delay on outputs with HW A/V sync 4733 if (usesHwAvSync()) { 4734 standbyDelay = 0; 4735 } else if (audio_is_linear_pcm(mFormat)) { 4736 standbyDelay = microseconds(activeSleepTime*2); 4737 } else { 4738 standbyDelay = kOffloadStandbyDelayNs; 4739 } 4740} 4741 4742void AudioFlinger::DirectOutputThread::flushHw_l() 4743{ 4744 mOutput->flush(); 4745 mHwPaused = false; 4746} 4747 4748// ---------------------------------------------------------------------------- 4749 4750AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4751 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4752 : Thread(false /*canCallJava*/), 4753 mPlaybackThread(playbackThread), 4754 mWriteAckSequence(0), 4755 mDrainSequence(0) 4756{ 4757} 4758 4759AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4760{ 4761} 4762 4763void AudioFlinger::AsyncCallbackThread::onFirstRef() 4764{ 4765 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4766} 4767 4768bool AudioFlinger::AsyncCallbackThread::threadLoop() 4769{ 4770 while (!exitPending()) { 4771 uint32_t writeAckSequence; 4772 uint32_t drainSequence; 4773 4774 { 4775 Mutex::Autolock _l(mLock); 4776 while (!((mWriteAckSequence & 1) || 4777 (mDrainSequence & 1) || 4778 exitPending())) { 4779 mWaitWorkCV.wait(mLock); 4780 } 4781 4782 if (exitPending()) { 4783 break; 4784 } 4785 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4786 mWriteAckSequence, mDrainSequence); 4787 writeAckSequence = mWriteAckSequence; 4788 mWriteAckSequence &= ~1; 4789 drainSequence = mDrainSequence; 4790 mDrainSequence &= ~1; 4791 } 4792 { 4793 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4794 if (playbackThread != 0) { 4795 if (writeAckSequence & 1) { 4796 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4797 } 4798 if (drainSequence & 1) { 4799 playbackThread->resetDraining(drainSequence >> 1); 4800 } 4801 } 4802 } 4803 } 4804 return false; 4805} 4806 4807void AudioFlinger::AsyncCallbackThread::exit() 4808{ 4809 ALOGV("AsyncCallbackThread::exit"); 4810 Mutex::Autolock _l(mLock); 4811 requestExit(); 4812 mWaitWorkCV.broadcast(); 4813} 4814 4815void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4816{ 4817 Mutex::Autolock _l(mLock); 4818 // bit 0 is cleared 4819 mWriteAckSequence = sequence << 1; 4820} 4821 4822void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4823{ 4824 Mutex::Autolock _l(mLock); 4825 // ignore unexpected callbacks 4826 if (mWriteAckSequence & 2) { 4827 mWriteAckSequence |= 1; 4828 mWaitWorkCV.signal(); 4829 } 4830} 4831 4832void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4833{ 4834 Mutex::Autolock _l(mLock); 4835 // bit 0 is cleared 4836 mDrainSequence = sequence << 1; 4837} 4838 4839void AudioFlinger::AsyncCallbackThread::resetDraining() 4840{ 4841 Mutex::Autolock _l(mLock); 4842 // ignore unexpected callbacks 4843 if (mDrainSequence & 2) { 4844 mDrainSequence |= 1; 4845 mWaitWorkCV.signal(); 4846 } 4847} 4848 4849 4850// ---------------------------------------------------------------------------- 4851AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4852 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 4853 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 4854 mPausedBytesRemaining(0) 4855{ 4856 //FIXME: mStandby should be set to true by ThreadBase constructor 4857 mStandby = true; 4858} 4859 4860void AudioFlinger::OffloadThread::threadLoop_exit() 4861{ 4862 if (mFlushPending || mHwPaused) { 4863 // If a flush is pending or track was paused, just discard buffered data 4864 flushHw_l(); 4865 } else { 4866 mMixerStatus = MIXER_DRAIN_ALL; 4867 threadLoop_drain(); 4868 } 4869 if (mUseAsyncWrite) { 4870 ALOG_ASSERT(mCallbackThread != 0); 4871 mCallbackThread->exit(); 4872 } 4873 PlaybackThread::threadLoop_exit(); 4874} 4875 4876AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4877 Vector< sp<Track> > *tracksToRemove 4878) 4879{ 4880 size_t count = mActiveTracks.size(); 4881 4882 mixer_state mixerStatus = MIXER_IDLE; 4883 bool doHwPause = false; 4884 bool doHwResume = false; 4885 4886 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4887 4888 // find out which tracks need to be processed 4889 for (size_t i = 0; i < count; i++) { 4890 sp<Track> t = mActiveTracks[i].promote(); 4891 // The track died recently 4892 if (t == 0) { 4893 continue; 4894 } 4895 Track* const track = t.get(); 4896 audio_track_cblk_t* cblk = track->cblk(); 4897 // Only consider last track started for volume and mixer state control. 4898 // In theory an older track could underrun and restart after the new one starts 4899 // but as we only care about the transition phase between two tracks on a 4900 // direct output, it is not a problem to ignore the underrun case. 4901 sp<Track> l = mLatestActiveTrack.promote(); 4902 bool last = l.get() == track; 4903 4904 if (track->isInvalid()) { 4905 ALOGW("An invalidated track shouldn't be in active list"); 4906 tracksToRemove->add(track); 4907 continue; 4908 } 4909 4910 if (track->mState == TrackBase::IDLE) { 4911 ALOGW("An idle track shouldn't be in active list"); 4912 continue; 4913 } 4914 4915 if (track->isPausing()) { 4916 track->setPaused(); 4917 if (last) { 4918 if (!mHwPaused) { 4919 doHwPause = true; 4920 mHwPaused = true; 4921 } 4922 // If we were part way through writing the mixbuffer to 4923 // the HAL we must save this until we resume 4924 // BUG - this will be wrong if a different track is made active, 4925 // in that case we want to discard the pending data in the 4926 // mixbuffer and tell the client to present it again when the 4927 // track is resumed 4928 mPausedWriteLength = mCurrentWriteLength; 4929 mPausedBytesRemaining = mBytesRemaining; 4930 mBytesRemaining = 0; // stop writing 4931 } 4932 tracksToRemove->add(track); 4933 } else if (track->isFlushPending()) { 4934 track->flushAck(); 4935 if (last) { 4936 mFlushPending = true; 4937 } 4938 } else if (track->isResumePending()){ 4939 track->resumeAck(); 4940 if (last) { 4941 if (mPausedBytesRemaining) { 4942 // Need to continue write that was interrupted 4943 mCurrentWriteLength = mPausedWriteLength; 4944 mBytesRemaining = mPausedBytesRemaining; 4945 mPausedBytesRemaining = 0; 4946 } 4947 if (mHwPaused) { 4948 doHwResume = true; 4949 mHwPaused = false; 4950 // threadLoop_mix() will handle the case that we need to 4951 // resume an interrupted write 4952 } 4953 // enable write to audio HAL 4954 sleepTime = 0; 4955 4956 // Do not handle new data in this iteration even if track->framesReady() 4957 mixerStatus = MIXER_TRACKS_ENABLED; 4958 } 4959 } else if (track->framesReady() && track->isReady() && 4960 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4961 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4962 if (track->mFillingUpStatus == Track::FS_FILLED) { 4963 track->mFillingUpStatus = Track::FS_ACTIVE; 4964 // make sure processVolume_l() will apply new volume even if 0 4965 mLeftVolFloat = mRightVolFloat = -1.0; 4966 } 4967 4968 if (last) { 4969 sp<Track> previousTrack = mPreviousTrack.promote(); 4970 if (previousTrack != 0) { 4971 if (track != previousTrack.get()) { 4972 // Flush any data still being written from last track 4973 mBytesRemaining = 0; 4974 if (mPausedBytesRemaining) { 4975 // Last track was paused so we also need to flush saved 4976 // mixbuffer state and invalidate track so that it will 4977 // re-submit that unwritten data when it is next resumed 4978 mPausedBytesRemaining = 0; 4979 // Invalidate is a bit drastic - would be more efficient 4980 // to have a flag to tell client that some of the 4981 // previously written data was lost 4982 previousTrack->invalidate(); 4983 } 4984 // flush data already sent to the DSP if changing audio session as audio 4985 // comes from a different source. Also invalidate previous track to force a 4986 // seek when resuming. 4987 if (previousTrack->sessionId() != track->sessionId()) { 4988 previousTrack->invalidate(); 4989 } 4990 } 4991 } 4992 mPreviousTrack = track; 4993 // reset retry count 4994 track->mRetryCount = kMaxTrackRetriesOffload; 4995 mActiveTrack = t; 4996 mixerStatus = MIXER_TRACKS_READY; 4997 } 4998 } else { 4999 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5000 if (track->isStopping_1()) { 5001 // Hardware buffer can hold a large amount of audio so we must 5002 // wait for all current track's data to drain before we say 5003 // that the track is stopped. 5004 if (mBytesRemaining == 0) { 5005 // Only start draining when all data in mixbuffer 5006 // has been written 5007 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5008 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5009 // do not drain if no data was ever sent to HAL (mStandby == true) 5010 if (last && !mStandby) { 5011 // do not modify drain sequence if we are already draining. This happens 5012 // when resuming from pause after drain. 5013 if ((mDrainSequence & 1) == 0) { 5014 sleepTime = 0; 5015 standbyTime = systemTime() + standbyDelay; 5016 mixerStatus = MIXER_DRAIN_TRACK; 5017 mDrainSequence += 2; 5018 } 5019 if (mHwPaused) { 5020 // It is possible to move from PAUSED to STOPPING_1 without 5021 // a resume so we must ensure hardware is running 5022 doHwResume = true; 5023 mHwPaused = false; 5024 } 5025 } 5026 } 5027 } else if (track->isStopping_2()) { 5028 // Drain has completed or we are in standby, signal presentation complete 5029 if (!(mDrainSequence & 1) || !last || mStandby) { 5030 track->mState = TrackBase::STOPPED; 5031 size_t audioHALFrames = 5032 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5033 size_t framesWritten = 5034 mBytesWritten / mOutput->getFrameSize(); 5035 track->presentationComplete(framesWritten, audioHALFrames); 5036 track->reset(); 5037 tracksToRemove->add(track); 5038 } 5039 } else { 5040 // No buffers for this track. Give it a few chances to 5041 // fill a buffer, then remove it from active list. 5042 if (--(track->mRetryCount) <= 0) { 5043 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5044 track->name()); 5045 tracksToRemove->add(track); 5046 // indicate to client process that the track was disabled because of underrun; 5047 // it will then automatically call start() when data is available 5048 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5049 } else if (last){ 5050 mixerStatus = MIXER_TRACKS_ENABLED; 5051 } 5052 } 5053 } 5054 // compute volume for this track 5055 processVolume_l(track, last); 5056 } 5057 5058 // make sure the pause/flush/resume sequence is executed in the right order. 5059 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5060 // before flush and then resume HW. This can happen in case of pause/flush/resume 5061 // if resume is received before pause is executed. 5062 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5063 mOutput->stream->pause(mOutput->stream); 5064 } 5065 if (mFlushPending) { 5066 flushHw_l(); 5067 mFlushPending = false; 5068 } 5069 if (!mStandby && doHwResume) { 5070 mOutput->stream->resume(mOutput->stream); 5071 } 5072 5073 // remove all the tracks that need to be... 5074 removeTracks_l(*tracksToRemove); 5075 5076 return mixerStatus; 5077} 5078 5079// must be called with thread mutex locked 5080bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5081{ 5082 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5083 mWriteAckSequence, mDrainSequence); 5084 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5085 return true; 5086 } 5087 return false; 5088} 5089 5090bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5091{ 5092 Mutex::Autolock _l(mLock); 5093 return waitingAsyncCallback_l(); 5094} 5095 5096void AudioFlinger::OffloadThread::flushHw_l() 5097{ 5098 DirectOutputThread::flushHw_l(); 5099 // Flush anything still waiting in the mixbuffer 5100 mCurrentWriteLength = 0; 5101 mBytesRemaining = 0; 5102 mPausedWriteLength = 0; 5103 mPausedBytesRemaining = 0; 5104 5105 if (mUseAsyncWrite) { 5106 // discard any pending drain or write ack by incrementing sequence 5107 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5108 mDrainSequence = (mDrainSequence + 2) & ~1; 5109 ALOG_ASSERT(mCallbackThread != 0); 5110 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5111 mCallbackThread->setDraining(mDrainSequence); 5112 } 5113} 5114 5115void AudioFlinger::OffloadThread::onAddNewTrack_l() 5116{ 5117 sp<Track> previousTrack = mPreviousTrack.promote(); 5118 sp<Track> latestTrack = mLatestActiveTrack.promote(); 5119 5120 if (previousTrack != 0 && latestTrack != 0 && 5121 (previousTrack->sessionId() != latestTrack->sessionId())) { 5122 mFlushPending = true; 5123 } 5124 PlaybackThread::onAddNewTrack_l(); 5125} 5126 5127// ---------------------------------------------------------------------------- 5128 5129AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5130 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5131 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5132 systemReady, DUPLICATING), 5133 mWaitTimeMs(UINT_MAX) 5134{ 5135 addOutputTrack(mainThread); 5136} 5137 5138AudioFlinger::DuplicatingThread::~DuplicatingThread() 5139{ 5140 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5141 mOutputTracks[i]->destroy(); 5142 } 5143} 5144 5145void AudioFlinger::DuplicatingThread::threadLoop_mix() 5146{ 5147 // mix buffers... 5148 if (outputsReady(outputTracks)) { 5149 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5150 } else { 5151 if (mMixerBufferValid) { 5152 memset(mMixerBuffer, 0, mMixerBufferSize); 5153 } else { 5154 memset(mSinkBuffer, 0, mSinkBufferSize); 5155 } 5156 } 5157 sleepTime = 0; 5158 writeFrames = mNormalFrameCount; 5159 mCurrentWriteLength = mSinkBufferSize; 5160 standbyTime = systemTime() + standbyDelay; 5161} 5162 5163void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5164{ 5165 if (sleepTime == 0) { 5166 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5167 sleepTime = activeSleepTime; 5168 } else { 5169 sleepTime = idleSleepTime; 5170 } 5171 } else if (mBytesWritten != 0) { 5172 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5173 writeFrames = mNormalFrameCount; 5174 memset(mSinkBuffer, 0, mSinkBufferSize); 5175 } else { 5176 // flush remaining overflow buffers in output tracks 5177 writeFrames = 0; 5178 } 5179 sleepTime = 0; 5180 } 5181} 5182 5183ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5184{ 5185 for (size_t i = 0; i < outputTracks.size(); i++) { 5186 outputTracks[i]->write(mSinkBuffer, writeFrames); 5187 } 5188 mStandby = false; 5189 return (ssize_t)mSinkBufferSize; 5190} 5191 5192void AudioFlinger::DuplicatingThread::threadLoop_standby() 5193{ 5194 // DuplicatingThread implements standby by stopping all tracks 5195 for (size_t i = 0; i < outputTracks.size(); i++) { 5196 outputTracks[i]->stop(); 5197 } 5198} 5199 5200void AudioFlinger::DuplicatingThread::saveOutputTracks() 5201{ 5202 outputTracks = mOutputTracks; 5203} 5204 5205void AudioFlinger::DuplicatingThread::clearOutputTracks() 5206{ 5207 outputTracks.clear(); 5208} 5209 5210void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5211{ 5212 Mutex::Autolock _l(mLock); 5213 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5214 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5215 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5216 const size_t frameCount = 5217 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5218 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5219 // from different OutputTracks and their associated MixerThreads (e.g. one may 5220 // nearly empty and the other may be dropping data). 5221 5222 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5223 this, 5224 mSampleRate, 5225 mFormat, 5226 mChannelMask, 5227 frameCount, 5228 IPCThreadState::self()->getCallingUid()); 5229 if (outputTrack->cblk() != NULL) { 5230 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5231 mOutputTracks.add(outputTrack); 5232 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5233 updateWaitTime_l(); 5234 } 5235} 5236 5237void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5238{ 5239 Mutex::Autolock _l(mLock); 5240 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5241 if (mOutputTracks[i]->thread() == thread) { 5242 mOutputTracks[i]->destroy(); 5243 mOutputTracks.removeAt(i); 5244 updateWaitTime_l(); 5245 if (thread->getOutput() == mOutput) { 5246 mOutput = NULL; 5247 } 5248 return; 5249 } 5250 } 5251 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5252} 5253 5254// caller must hold mLock 5255void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5256{ 5257 mWaitTimeMs = UINT_MAX; 5258 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5259 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5260 if (strong != 0) { 5261 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5262 if (waitTimeMs < mWaitTimeMs) { 5263 mWaitTimeMs = waitTimeMs; 5264 } 5265 } 5266 } 5267} 5268 5269 5270bool AudioFlinger::DuplicatingThread::outputsReady( 5271 const SortedVector< sp<OutputTrack> > &outputTracks) 5272{ 5273 for (size_t i = 0; i < outputTracks.size(); i++) { 5274 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5275 if (thread == 0) { 5276 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5277 outputTracks[i].get()); 5278 return false; 5279 } 5280 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5281 // see note at standby() declaration 5282 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5283 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5284 thread.get()); 5285 return false; 5286 } 5287 } 5288 return true; 5289} 5290 5291uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5292{ 5293 return (mWaitTimeMs * 1000) / 2; 5294} 5295 5296void AudioFlinger::DuplicatingThread::cacheParameters_l() 5297{ 5298 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5299 updateWaitTime_l(); 5300 5301 MixerThread::cacheParameters_l(); 5302} 5303 5304// ---------------------------------------------------------------------------- 5305// Record 5306// ---------------------------------------------------------------------------- 5307 5308AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5309 AudioStreamIn *input, 5310 audio_io_handle_t id, 5311 audio_devices_t outDevice, 5312 audio_devices_t inDevice, 5313 bool systemReady 5314#ifdef TEE_SINK 5315 , const sp<NBAIO_Sink>& teeSink 5316#endif 5317 ) : 5318 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5319 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5320 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5321 mRsmpInRear(0) 5322#ifdef TEE_SINK 5323 , mTeeSink(teeSink) 5324#endif 5325 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5326 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5327 // mFastCapture below 5328 , mFastCaptureFutex(0) 5329 // mInputSource 5330 // mPipeSink 5331 // mPipeSource 5332 , mPipeFramesP2(0) 5333 // mPipeMemory 5334 // mFastCaptureNBLogWriter 5335 , mFastTrackAvail(false) 5336{ 5337 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5338 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5339 5340 readInputParameters_l(); 5341 5342 // create an NBAIO source for the HAL input stream, and negotiate 5343 mInputSource = new AudioStreamInSource(input->stream); 5344 size_t numCounterOffers = 0; 5345 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5346 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5347 ALOG_ASSERT(index == 0); 5348 5349 // initialize fast capture depending on configuration 5350 bool initFastCapture; 5351 switch (kUseFastCapture) { 5352 case FastCapture_Never: 5353 initFastCapture = false; 5354 break; 5355 case FastCapture_Always: 5356 initFastCapture = true; 5357 break; 5358 case FastCapture_Static: 5359 uint32_t primaryOutputSampleRate; 5360 { 5361 AutoMutex _l(audioFlinger->mHardwareLock); 5362 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5363 } 5364 initFastCapture = 5365 // either capture sample rate is same as (a reasonable) primary output sample rate 5366 ((isMusicRate(primaryOutputSampleRate) && 5367 (mSampleRate == primaryOutputSampleRate)) || 5368 // or primary output sample rate is unknown, and capture sample rate is reasonable 5369 ((primaryOutputSampleRate == 0) && 5370 isMusicRate(mSampleRate))) && 5371 // and the buffer size is < 12 ms 5372 (mFrameCount * 1000) / mSampleRate < 12; 5373 break; 5374 // case FastCapture_Dynamic: 5375 } 5376 5377 if (initFastCapture) { 5378 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5379 NBAIO_Format format = mInputSource->format(); 5380 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5381 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5382 void *pipeBuffer; 5383 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5384 sp<IMemory> pipeMemory; 5385 if ((roHeap == 0) || 5386 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5387 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5388 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5389 goto failed; 5390 } 5391 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5392 memset(pipeBuffer, 0, pipeSize); 5393 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5394 const NBAIO_Format offers[1] = {format}; 5395 size_t numCounterOffers = 0; 5396 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5397 ALOG_ASSERT(index == 0); 5398 mPipeSink = pipe; 5399 PipeReader *pipeReader = new PipeReader(*pipe); 5400 numCounterOffers = 0; 5401 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5402 ALOG_ASSERT(index == 0); 5403 mPipeSource = pipeReader; 5404 mPipeFramesP2 = pipeFramesP2; 5405 mPipeMemory = pipeMemory; 5406 5407 // create fast capture 5408 mFastCapture = new FastCapture(); 5409 FastCaptureStateQueue *sq = mFastCapture->sq(); 5410#ifdef STATE_QUEUE_DUMP 5411 // FIXME 5412#endif 5413 FastCaptureState *state = sq->begin(); 5414 state->mCblk = NULL; 5415 state->mInputSource = mInputSource.get(); 5416 state->mInputSourceGen++; 5417 state->mPipeSink = pipe; 5418 state->mPipeSinkGen++; 5419 state->mFrameCount = mFrameCount; 5420 state->mCommand = FastCaptureState::COLD_IDLE; 5421 // already done in constructor initialization list 5422 //mFastCaptureFutex = 0; 5423 state->mColdFutexAddr = &mFastCaptureFutex; 5424 state->mColdGen++; 5425 state->mDumpState = &mFastCaptureDumpState; 5426#ifdef TEE_SINK 5427 // FIXME 5428#endif 5429 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5430 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5431 sq->end(); 5432 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5433 5434 // start the fast capture 5435 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5436 pid_t tid = mFastCapture->getTid(); 5437 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5438#ifdef AUDIO_WATCHDOG 5439 // FIXME 5440#endif 5441 5442 mFastTrackAvail = true; 5443 } 5444failed: ; 5445 5446 // FIXME mNormalSource 5447} 5448 5449AudioFlinger::RecordThread::~RecordThread() 5450{ 5451 if (mFastCapture != 0) { 5452 FastCaptureStateQueue *sq = mFastCapture->sq(); 5453 FastCaptureState *state = sq->begin(); 5454 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5455 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5456 if (old == -1) { 5457 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5458 } 5459 } 5460 state->mCommand = FastCaptureState::EXIT; 5461 sq->end(); 5462 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5463 mFastCapture->join(); 5464 mFastCapture.clear(); 5465 } 5466 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5467 mAudioFlinger->unregisterWriter(mNBLogWriter); 5468 free(mRsmpInBuffer); 5469} 5470 5471void AudioFlinger::RecordThread::onFirstRef() 5472{ 5473 run(mThreadName, PRIORITY_URGENT_AUDIO); 5474} 5475 5476bool AudioFlinger::RecordThread::threadLoop() 5477{ 5478 nsecs_t lastWarning = 0; 5479 5480 inputStandBy(); 5481 5482reacquire_wakelock: 5483 sp<RecordTrack> activeTrack; 5484 int activeTracksGen; 5485 { 5486 Mutex::Autolock _l(mLock); 5487 size_t size = mActiveTracks.size(); 5488 activeTracksGen = mActiveTracksGen; 5489 if (size > 0) { 5490 // FIXME an arbitrary choice 5491 activeTrack = mActiveTracks[0]; 5492 acquireWakeLock_l(activeTrack->uid()); 5493 if (size > 1) { 5494 SortedVector<int> tmp; 5495 for (size_t i = 0; i < size; i++) { 5496 tmp.add(mActiveTracks[i]->uid()); 5497 } 5498 updateWakeLockUids_l(tmp); 5499 } 5500 } else { 5501 acquireWakeLock_l(-1); 5502 } 5503 } 5504 5505 // used to request a deferred sleep, to be executed later while mutex is unlocked 5506 uint32_t sleepUs = 0; 5507 5508 // loop while there is work to do 5509 for (;;) { 5510 Vector< sp<EffectChain> > effectChains; 5511 5512 // sleep with mutex unlocked 5513 if (sleepUs > 0) { 5514 ATRACE_BEGIN("sleep"); 5515 usleep(sleepUs); 5516 ATRACE_END(); 5517 sleepUs = 0; 5518 } 5519 5520 // activeTracks accumulates a copy of a subset of mActiveTracks 5521 Vector< sp<RecordTrack> > activeTracks; 5522 5523 // reference to the (first and only) active fast track 5524 sp<RecordTrack> fastTrack; 5525 5526 // reference to a fast track which is about to be removed 5527 sp<RecordTrack> fastTrackToRemove; 5528 5529 { // scope for mLock 5530 Mutex::Autolock _l(mLock); 5531 5532 processConfigEvents_l(); 5533 5534 // check exitPending here because checkForNewParameters_l() and 5535 // checkForNewParameters_l() can temporarily release mLock 5536 if (exitPending()) { 5537 break; 5538 } 5539 5540 // if no active track(s), then standby and release wakelock 5541 size_t size = mActiveTracks.size(); 5542 if (size == 0) { 5543 standbyIfNotAlreadyInStandby(); 5544 // exitPending() can't become true here 5545 releaseWakeLock_l(); 5546 ALOGV("RecordThread: loop stopping"); 5547 // go to sleep 5548 mWaitWorkCV.wait(mLock); 5549 ALOGV("RecordThread: loop starting"); 5550 goto reacquire_wakelock; 5551 } 5552 5553 if (mActiveTracksGen != activeTracksGen) { 5554 activeTracksGen = mActiveTracksGen; 5555 SortedVector<int> tmp; 5556 for (size_t i = 0; i < size; i++) { 5557 tmp.add(mActiveTracks[i]->uid()); 5558 } 5559 updateWakeLockUids_l(tmp); 5560 } 5561 5562 bool doBroadcast = false; 5563 for (size_t i = 0; i < size; ) { 5564 5565 activeTrack = mActiveTracks[i]; 5566 if (activeTrack->isTerminated()) { 5567 if (activeTrack->isFastTrack()) { 5568 ALOG_ASSERT(fastTrackToRemove == 0); 5569 fastTrackToRemove = activeTrack; 5570 } 5571 removeTrack_l(activeTrack); 5572 mActiveTracks.remove(activeTrack); 5573 mActiveTracksGen++; 5574 size--; 5575 continue; 5576 } 5577 5578 TrackBase::track_state activeTrackState = activeTrack->mState; 5579 switch (activeTrackState) { 5580 5581 case TrackBase::PAUSING: 5582 mActiveTracks.remove(activeTrack); 5583 mActiveTracksGen++; 5584 doBroadcast = true; 5585 size--; 5586 continue; 5587 5588 case TrackBase::STARTING_1: 5589 sleepUs = 10000; 5590 i++; 5591 continue; 5592 5593 case TrackBase::STARTING_2: 5594 doBroadcast = true; 5595 mStandby = false; 5596 activeTrack->mState = TrackBase::ACTIVE; 5597 break; 5598 5599 case TrackBase::ACTIVE: 5600 break; 5601 5602 case TrackBase::IDLE: 5603 i++; 5604 continue; 5605 5606 default: 5607 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5608 } 5609 5610 activeTracks.add(activeTrack); 5611 i++; 5612 5613 if (activeTrack->isFastTrack()) { 5614 ALOG_ASSERT(!mFastTrackAvail); 5615 ALOG_ASSERT(fastTrack == 0); 5616 fastTrack = activeTrack; 5617 } 5618 } 5619 if (doBroadcast) { 5620 mStartStopCond.broadcast(); 5621 } 5622 5623 // sleep if there are no active tracks to process 5624 if (activeTracks.size() == 0) { 5625 if (sleepUs == 0) { 5626 sleepUs = kRecordThreadSleepUs; 5627 } 5628 continue; 5629 } 5630 sleepUs = 0; 5631 5632 lockEffectChains_l(effectChains); 5633 } 5634 5635 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5636 5637 size_t size = effectChains.size(); 5638 for (size_t i = 0; i < size; i++) { 5639 // thread mutex is not locked, but effect chain is locked 5640 effectChains[i]->process_l(); 5641 } 5642 5643 // Push a new fast capture state if fast capture is not already running, or cblk change 5644 if (mFastCapture != 0) { 5645 FastCaptureStateQueue *sq = mFastCapture->sq(); 5646 FastCaptureState *state = sq->begin(); 5647 bool didModify = false; 5648 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5649 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5650 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5651 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5652 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5653 if (old == -1) { 5654 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5655 } 5656 } 5657 state->mCommand = FastCaptureState::READ_WRITE; 5658#if 0 // FIXME 5659 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5660 FastThreadDumpState::kSamplingNforLowRamDevice : 5661 FastThreadDumpState::kSamplingN); 5662#endif 5663 didModify = true; 5664 } 5665 audio_track_cblk_t *cblkOld = state->mCblk; 5666 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5667 if (cblkNew != cblkOld) { 5668 state->mCblk = cblkNew; 5669 // block until acked if removing a fast track 5670 if (cblkOld != NULL) { 5671 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5672 } 5673 didModify = true; 5674 } 5675 sq->end(didModify); 5676 if (didModify) { 5677 sq->push(block); 5678#if 0 5679 if (kUseFastCapture == FastCapture_Dynamic) { 5680 mNormalSource = mPipeSource; 5681 } 5682#endif 5683 } 5684 } 5685 5686 // now run the fast track destructor with thread mutex unlocked 5687 fastTrackToRemove.clear(); 5688 5689 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5690 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5691 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5692 // If destination is non-contiguous, first read past the nominal end of buffer, then 5693 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5694 5695 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5696 ssize_t framesRead; 5697 5698 // If an NBAIO source is present, use it to read the normal capture's data 5699 if (mPipeSource != 0) { 5700 size_t framesToRead = mBufferSize / mFrameSize; 5701 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5702 framesToRead, AudioBufferProvider::kInvalidPTS); 5703 if (framesRead == 0) { 5704 // since pipe is non-blocking, simulate blocking input 5705 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5706 } 5707 // otherwise use the HAL / AudioStreamIn directly 5708 } else { 5709 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5710 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5711 if (bytesRead < 0) { 5712 framesRead = bytesRead; 5713 } else { 5714 framesRead = bytesRead / mFrameSize; 5715 } 5716 } 5717 5718 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5719 ALOGE("read failed: framesRead=%d", framesRead); 5720 // Force input into standby so that it tries to recover at next read attempt 5721 inputStandBy(); 5722 sleepUs = kRecordThreadSleepUs; 5723 } 5724 if (framesRead <= 0) { 5725 goto unlock; 5726 } 5727 ALOG_ASSERT(framesRead > 0); 5728 5729 if (mTeeSink != 0) { 5730 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5731 } 5732 // If destination is non-contiguous, we now correct for reading past end of buffer. 5733 { 5734 size_t part1 = mRsmpInFramesP2 - rear; 5735 if ((size_t) framesRead > part1) { 5736 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5737 (framesRead - part1) * mFrameSize); 5738 } 5739 } 5740 rear = mRsmpInRear += framesRead; 5741 5742 size = activeTracks.size(); 5743 // loop over each active track 5744 for (size_t i = 0; i < size; i++) { 5745 activeTrack = activeTracks[i]; 5746 5747 // skip fast tracks, as those are handled directly by FastCapture 5748 if (activeTrack->isFastTrack()) { 5749 continue; 5750 } 5751 5752 // TODO: This code probably should be moved to RecordTrack. 5753 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5754 5755 enum { 5756 OVERRUN_UNKNOWN, 5757 OVERRUN_TRUE, 5758 OVERRUN_FALSE 5759 } overrun = OVERRUN_UNKNOWN; 5760 5761 // loop over getNextBuffer to handle circular sink 5762 for (;;) { 5763 5764 activeTrack->mSink.frameCount = ~0; 5765 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5766 size_t framesOut = activeTrack->mSink.frameCount; 5767 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5768 5769 // check available frames and handle overrun conditions 5770 // if the record track isn't draining fast enough. 5771 bool hasOverrun; 5772 size_t framesIn; 5773 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5774 if (hasOverrun) { 5775 overrun = OVERRUN_TRUE; 5776 } 5777 if (framesOut == 0 || framesIn == 0) { 5778 break; 5779 } 5780 5781 // Don't allow framesOut to be larger than what is possible with resampling 5782 // from framesIn. 5783 // This isn't strictly necessary but helps limit buffer resizing in 5784 // RecordBufferConverter. TODO: remove when no longer needed. 5785 framesOut = min(framesOut, 5786 destinationFramesPossible( 5787 framesIn, mSampleRate, activeTrack->mSampleRate)); 5788 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5789 framesOut = activeTrack->mRecordBufferConverter->convert( 5790 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5791 5792 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5793 overrun = OVERRUN_FALSE; 5794 } 5795 5796 if (activeTrack->mFramesToDrop == 0) { 5797 if (framesOut > 0) { 5798 activeTrack->mSink.frameCount = framesOut; 5799 activeTrack->releaseBuffer(&activeTrack->mSink); 5800 } 5801 } else { 5802 // FIXME could do a partial drop of framesOut 5803 if (activeTrack->mFramesToDrop > 0) { 5804 activeTrack->mFramesToDrop -= framesOut; 5805 if (activeTrack->mFramesToDrop <= 0) { 5806 activeTrack->clearSyncStartEvent(); 5807 } 5808 } else { 5809 activeTrack->mFramesToDrop += framesOut; 5810 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5811 activeTrack->mSyncStartEvent->isCancelled()) { 5812 ALOGW("Synced record %s, session %d, trigger session %d", 5813 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5814 activeTrack->sessionId(), 5815 (activeTrack->mSyncStartEvent != 0) ? 5816 activeTrack->mSyncStartEvent->triggerSession() : 0); 5817 activeTrack->clearSyncStartEvent(); 5818 } 5819 } 5820 } 5821 5822 if (framesOut == 0) { 5823 break; 5824 } 5825 } 5826 5827 switch (overrun) { 5828 case OVERRUN_TRUE: 5829 // client isn't retrieving buffers fast enough 5830 if (!activeTrack->setOverflow()) { 5831 nsecs_t now = systemTime(); 5832 // FIXME should lastWarning per track? 5833 if ((now - lastWarning) > kWarningThrottleNs) { 5834 ALOGW("RecordThread: buffer overflow"); 5835 lastWarning = now; 5836 } 5837 } 5838 break; 5839 case OVERRUN_FALSE: 5840 activeTrack->clearOverflow(); 5841 break; 5842 case OVERRUN_UNKNOWN: 5843 break; 5844 } 5845 5846 } 5847 5848unlock: 5849 // enable changes in effect chain 5850 unlockEffectChains(effectChains); 5851 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5852 } 5853 5854 standbyIfNotAlreadyInStandby(); 5855 5856 { 5857 Mutex::Autolock _l(mLock); 5858 for (size_t i = 0; i < mTracks.size(); i++) { 5859 sp<RecordTrack> track = mTracks[i]; 5860 track->invalidate(); 5861 } 5862 mActiveTracks.clear(); 5863 mActiveTracksGen++; 5864 mStartStopCond.broadcast(); 5865 } 5866 5867 releaseWakeLock(); 5868 5869 ALOGV("RecordThread %p exiting", this); 5870 return false; 5871} 5872 5873void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5874{ 5875 if (!mStandby) { 5876 inputStandBy(); 5877 mStandby = true; 5878 } 5879} 5880 5881void AudioFlinger::RecordThread::inputStandBy() 5882{ 5883 // Idle the fast capture if it's currently running 5884 if (mFastCapture != 0) { 5885 FastCaptureStateQueue *sq = mFastCapture->sq(); 5886 FastCaptureState *state = sq->begin(); 5887 if (!(state->mCommand & FastCaptureState::IDLE)) { 5888 state->mCommand = FastCaptureState::COLD_IDLE; 5889 state->mColdFutexAddr = &mFastCaptureFutex; 5890 state->mColdGen++; 5891 mFastCaptureFutex = 0; 5892 sq->end(); 5893 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5894 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5895#if 0 5896 if (kUseFastCapture == FastCapture_Dynamic) { 5897 // FIXME 5898 } 5899#endif 5900#ifdef AUDIO_WATCHDOG 5901 // FIXME 5902#endif 5903 } else { 5904 sq->end(false /*didModify*/); 5905 } 5906 } 5907 mInput->stream->common.standby(&mInput->stream->common); 5908} 5909 5910// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5911sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5912 const sp<AudioFlinger::Client>& client, 5913 uint32_t sampleRate, 5914 audio_format_t format, 5915 audio_channel_mask_t channelMask, 5916 size_t *pFrameCount, 5917 int sessionId, 5918 size_t *notificationFrames, 5919 int uid, 5920 IAudioFlinger::track_flags_t *flags, 5921 pid_t tid, 5922 status_t *status) 5923{ 5924 size_t frameCount = *pFrameCount; 5925 sp<RecordTrack> track; 5926 status_t lStatus; 5927 5928 // client expresses a preference for FAST, but we get the final say 5929 if (*flags & IAudioFlinger::TRACK_FAST) { 5930 if ( 5931 // we formerly checked for a callback handler (non-0 tid), 5932 // but that is no longer required for TRANSFER_OBTAIN mode 5933 // 5934 // frame count is not specified, or is exactly the pipe depth 5935 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5936 // PCM data 5937 audio_is_linear_pcm(format) && 5938 // native format 5939 (format == mFormat) && 5940 // native channel mask 5941 (channelMask == mChannelMask) && 5942 // native hardware sample rate 5943 (sampleRate == mSampleRate) && 5944 // record thread has an associated fast capture 5945 hasFastCapture() && 5946 // there are sufficient fast track slots available 5947 mFastTrackAvail 5948 ) { 5949 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5950 frameCount, mFrameCount); 5951 } else { 5952 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5953 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5954 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5955 frameCount, mFrameCount, mPipeFramesP2, 5956 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5957 hasFastCapture(), tid, mFastTrackAvail); 5958 *flags &= ~IAudioFlinger::TRACK_FAST; 5959 } 5960 } 5961 5962 // compute track buffer size in frames, and suggest the notification frame count 5963 if (*flags & IAudioFlinger::TRACK_FAST) { 5964 // fast track: frame count is exactly the pipe depth 5965 frameCount = mPipeFramesP2; 5966 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5967 *notificationFrames = mFrameCount; 5968 } else { 5969 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5970 // or 20 ms if there is a fast capture 5971 // TODO This could be a roundupRatio inline, and const 5972 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5973 * sampleRate + mSampleRate - 1) / mSampleRate; 5974 // minimum number of notification periods is at least kMinNotifications, 5975 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5976 static const size_t kMinNotifications = 3; 5977 static const uint32_t kMinMs = 30; 5978 // TODO This could be a roundupRatio inline 5979 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5980 // TODO This could be a roundupRatio inline 5981 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5982 maxNotificationFrames; 5983 const size_t minFrameCount = maxNotificationFrames * 5984 max(kMinNotifications, minNotificationsByMs); 5985 frameCount = max(frameCount, minFrameCount); 5986 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5987 *notificationFrames = maxNotificationFrames; 5988 } 5989 } 5990 *pFrameCount = frameCount; 5991 5992 lStatus = initCheck(); 5993 if (lStatus != NO_ERROR) { 5994 ALOGE("createRecordTrack_l() audio driver not initialized"); 5995 goto Exit; 5996 } 5997 5998 { // scope for mLock 5999 Mutex::Autolock _l(mLock); 6000 6001 track = new RecordTrack(this, client, sampleRate, 6002 format, channelMask, frameCount, NULL, sessionId, uid, 6003 *flags, TrackBase::TYPE_DEFAULT); 6004 6005 lStatus = track->initCheck(); 6006 if (lStatus != NO_ERROR) { 6007 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6008 // track must be cleared from the caller as the caller has the AF lock 6009 goto Exit; 6010 } 6011 mTracks.add(track); 6012 6013 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6014 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6015 mAudioFlinger->btNrecIsOff(); 6016 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6017 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6018 6019 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6020 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6021 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6022 // so ask activity manager to do this on our behalf 6023 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6024 } 6025 } 6026 6027 lStatus = NO_ERROR; 6028 6029Exit: 6030 *status = lStatus; 6031 return track; 6032} 6033 6034status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6035 AudioSystem::sync_event_t event, 6036 int triggerSession) 6037{ 6038 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6039 sp<ThreadBase> strongMe = this; 6040 status_t status = NO_ERROR; 6041 6042 if (event == AudioSystem::SYNC_EVENT_NONE) { 6043 recordTrack->clearSyncStartEvent(); 6044 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6045 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6046 triggerSession, 6047 recordTrack->sessionId(), 6048 syncStartEventCallback, 6049 recordTrack); 6050 // Sync event can be cancelled by the trigger session if the track is not in a 6051 // compatible state in which case we start record immediately 6052 if (recordTrack->mSyncStartEvent->isCancelled()) { 6053 recordTrack->clearSyncStartEvent(); 6054 } else { 6055 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6056 recordTrack->mFramesToDrop = - 6057 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6058 } 6059 } 6060 6061 { 6062 // This section is a rendezvous between binder thread executing start() and RecordThread 6063 AutoMutex lock(mLock); 6064 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6065 if (recordTrack->mState == TrackBase::PAUSING) { 6066 ALOGV("active record track PAUSING -> ACTIVE"); 6067 recordTrack->mState = TrackBase::ACTIVE; 6068 } else { 6069 ALOGV("active record track state %d", recordTrack->mState); 6070 } 6071 return status; 6072 } 6073 6074 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6075 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6076 // or using a separate command thread 6077 recordTrack->mState = TrackBase::STARTING_1; 6078 mActiveTracks.add(recordTrack); 6079 mActiveTracksGen++; 6080 status_t status = NO_ERROR; 6081 if (recordTrack->isExternalTrack()) { 6082 mLock.unlock(); 6083 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6084 mLock.lock(); 6085 // FIXME should verify that recordTrack is still in mActiveTracks 6086 if (status != NO_ERROR) { 6087 mActiveTracks.remove(recordTrack); 6088 mActiveTracksGen++; 6089 recordTrack->clearSyncStartEvent(); 6090 ALOGV("RecordThread::start error %d", status); 6091 return status; 6092 } 6093 } 6094 // Catch up with current buffer indices if thread is already running. 6095 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6096 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6097 // see previously buffered data before it called start(), but with greater risk of overrun. 6098 6099 recordTrack->mResamplerBufferProvider->reset(); 6100 // clear any converter state as new data will be discontinuous 6101 recordTrack->mRecordBufferConverter->reset(); 6102 recordTrack->mState = TrackBase::STARTING_2; 6103 // signal thread to start 6104 mWaitWorkCV.broadcast(); 6105 if (mActiveTracks.indexOf(recordTrack) < 0) { 6106 ALOGV("Record failed to start"); 6107 status = BAD_VALUE; 6108 goto startError; 6109 } 6110 return status; 6111 } 6112 6113startError: 6114 if (recordTrack->isExternalTrack()) { 6115 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6116 } 6117 recordTrack->clearSyncStartEvent(); 6118 // FIXME I wonder why we do not reset the state here? 6119 return status; 6120} 6121 6122void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6123{ 6124 sp<SyncEvent> strongEvent = event.promote(); 6125 6126 if (strongEvent != 0) { 6127 sp<RefBase> ptr = strongEvent->cookie().promote(); 6128 if (ptr != 0) { 6129 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6130 recordTrack->handleSyncStartEvent(strongEvent); 6131 } 6132 } 6133} 6134 6135bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6136 ALOGV("RecordThread::stop"); 6137 AutoMutex _l(mLock); 6138 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6139 return false; 6140 } 6141 // note that threadLoop may still be processing the track at this point [without lock] 6142 recordTrack->mState = TrackBase::PAUSING; 6143 // do not wait for mStartStopCond if exiting 6144 if (exitPending()) { 6145 return true; 6146 } 6147 // FIXME incorrect usage of wait: no explicit predicate or loop 6148 mStartStopCond.wait(mLock); 6149 // if we have been restarted, recordTrack is in mActiveTracks here 6150 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6151 ALOGV("Record stopped OK"); 6152 return true; 6153 } 6154 return false; 6155} 6156 6157bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6158{ 6159 return false; 6160} 6161 6162status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6163{ 6164#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6165 if (!isValidSyncEvent(event)) { 6166 return BAD_VALUE; 6167 } 6168 6169 int eventSession = event->triggerSession(); 6170 status_t ret = NAME_NOT_FOUND; 6171 6172 Mutex::Autolock _l(mLock); 6173 6174 for (size_t i = 0; i < mTracks.size(); i++) { 6175 sp<RecordTrack> track = mTracks[i]; 6176 if (eventSession == track->sessionId()) { 6177 (void) track->setSyncEvent(event); 6178 ret = NO_ERROR; 6179 } 6180 } 6181 return ret; 6182#else 6183 return BAD_VALUE; 6184#endif 6185} 6186 6187// destroyTrack_l() must be called with ThreadBase::mLock held 6188void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6189{ 6190 track->terminate(); 6191 track->mState = TrackBase::STOPPED; 6192 // active tracks are removed by threadLoop() 6193 if (mActiveTracks.indexOf(track) < 0) { 6194 removeTrack_l(track); 6195 } 6196} 6197 6198void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6199{ 6200 mTracks.remove(track); 6201 // need anything related to effects here? 6202 if (track->isFastTrack()) { 6203 ALOG_ASSERT(!mFastTrackAvail); 6204 mFastTrackAvail = true; 6205 } 6206} 6207 6208void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6209{ 6210 dumpInternals(fd, args); 6211 dumpTracks(fd, args); 6212 dumpEffectChains(fd, args); 6213} 6214 6215void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6216{ 6217 dprintf(fd, "\nInput thread %p:\n", this); 6218 6219 dumpBase(fd, args); 6220 6221 if (mActiveTracks.size() == 0) { 6222 dprintf(fd, " No active record clients\n"); 6223 } 6224 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6225 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6226 6227 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6228 const FastCaptureDumpState copy(mFastCaptureDumpState); 6229 copy.dump(fd); 6230} 6231 6232void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6233{ 6234 const size_t SIZE = 256; 6235 char buffer[SIZE]; 6236 String8 result; 6237 6238 size_t numtracks = mTracks.size(); 6239 size_t numactive = mActiveTracks.size(); 6240 size_t numactiveseen = 0; 6241 dprintf(fd, " %d Tracks", numtracks); 6242 if (numtracks) { 6243 dprintf(fd, " of which %d are active\n", numactive); 6244 RecordTrack::appendDumpHeader(result); 6245 for (size_t i = 0; i < numtracks ; ++i) { 6246 sp<RecordTrack> track = mTracks[i]; 6247 if (track != 0) { 6248 bool active = mActiveTracks.indexOf(track) >= 0; 6249 if (active) { 6250 numactiveseen++; 6251 } 6252 track->dump(buffer, SIZE, active); 6253 result.append(buffer); 6254 } 6255 } 6256 } else { 6257 dprintf(fd, "\n"); 6258 } 6259 6260 if (numactiveseen != numactive) { 6261 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6262 " not in the track list\n"); 6263 result.append(buffer); 6264 RecordTrack::appendDumpHeader(result); 6265 for (size_t i = 0; i < numactive; ++i) { 6266 sp<RecordTrack> track = mActiveTracks[i]; 6267 if (mTracks.indexOf(track) < 0) { 6268 track->dump(buffer, SIZE, true); 6269 result.append(buffer); 6270 } 6271 } 6272 6273 } 6274 write(fd, result.string(), result.size()); 6275} 6276 6277 6278void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6279{ 6280 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6281 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6282 mRsmpInFront = recordThread->mRsmpInRear; 6283 mRsmpInUnrel = 0; 6284} 6285 6286void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6287 size_t *framesAvailable, bool *hasOverrun) 6288{ 6289 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6290 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6291 const int32_t rear = recordThread->mRsmpInRear; 6292 const int32_t front = mRsmpInFront; 6293 const ssize_t filled = rear - front; 6294 6295 size_t framesIn; 6296 bool overrun = false; 6297 if (filled < 0) { 6298 // should not happen, but treat like a massive overrun and re-sync 6299 framesIn = 0; 6300 mRsmpInFront = rear; 6301 overrun = true; 6302 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6303 framesIn = (size_t) filled; 6304 } else { 6305 // client is not keeping up with server, but give it latest data 6306 framesIn = recordThread->mRsmpInFrames; 6307 mRsmpInFront = /* front = */ rear - framesIn; 6308 overrun = true; 6309 } 6310 if (framesAvailable != NULL) { 6311 *framesAvailable = framesIn; 6312 } 6313 if (hasOverrun != NULL) { 6314 *hasOverrun = overrun; 6315 } 6316} 6317 6318// AudioBufferProvider interface 6319status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6320 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6321{ 6322 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6323 if (threadBase == 0) { 6324 buffer->frameCount = 0; 6325 buffer->raw = NULL; 6326 return NOT_ENOUGH_DATA; 6327 } 6328 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6329 int32_t rear = recordThread->mRsmpInRear; 6330 int32_t front = mRsmpInFront; 6331 ssize_t filled = rear - front; 6332 // FIXME should not be P2 (don't want to increase latency) 6333 // FIXME if client not keeping up, discard 6334 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6335 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6336 front &= recordThread->mRsmpInFramesP2 - 1; 6337 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6338 if (part1 > (size_t) filled) { 6339 part1 = filled; 6340 } 6341 size_t ask = buffer->frameCount; 6342 ALOG_ASSERT(ask > 0); 6343 if (part1 > ask) { 6344 part1 = ask; 6345 } 6346 if (part1 == 0) { 6347 // out of data is fine since the resampler will return a short-count. 6348 buffer->raw = NULL; 6349 buffer->frameCount = 0; 6350 mRsmpInUnrel = 0; 6351 return NOT_ENOUGH_DATA; 6352 } 6353 6354 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6355 buffer->frameCount = part1; 6356 mRsmpInUnrel = part1; 6357 return NO_ERROR; 6358} 6359 6360// AudioBufferProvider interface 6361void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6362 AudioBufferProvider::Buffer* buffer) 6363{ 6364 size_t stepCount = buffer->frameCount; 6365 if (stepCount == 0) { 6366 return; 6367 } 6368 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6369 mRsmpInUnrel -= stepCount; 6370 mRsmpInFront += stepCount; 6371 buffer->raw = NULL; 6372 buffer->frameCount = 0; 6373} 6374 6375AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6376 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6377 uint32_t srcSampleRate, 6378 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6379 uint32_t dstSampleRate) : 6380 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6381 // mSrcFormat 6382 // mSrcSampleRate 6383 // mDstChannelMask 6384 // mDstFormat 6385 // mDstSampleRate 6386 // mSrcChannelCount 6387 // mDstChannelCount 6388 // mDstFrameSize 6389 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6390 mResampler(NULL), 6391 mIsLegacyDownmix(false), 6392 mIsLegacyUpmix(false), 6393 mRequiresFloat(false), 6394 mInputConverterProvider(NULL) 6395{ 6396 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6397 dstChannelMask, dstFormat, dstSampleRate); 6398} 6399 6400AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6401 free(mBuf); 6402 delete mResampler; 6403 delete mInputConverterProvider; 6404} 6405 6406size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6407 AudioBufferProvider *provider, size_t frames) 6408{ 6409 if (mInputConverterProvider != NULL) { 6410 mInputConverterProvider->setBufferProvider(provider); 6411 provider = mInputConverterProvider; 6412 } 6413 6414 if (mResampler == NULL) { 6415 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6416 mSrcSampleRate, mSrcFormat, mDstFormat); 6417 6418 AudioBufferProvider::Buffer buffer; 6419 for (size_t i = frames; i > 0; ) { 6420 buffer.frameCount = i; 6421 status_t status = provider->getNextBuffer(&buffer, 0); 6422 if (status != OK || buffer.frameCount == 0) { 6423 frames -= i; // cannot fill request. 6424 break; 6425 } 6426 // format convert to destination buffer 6427 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6428 6429 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6430 i -= buffer.frameCount; 6431 provider->releaseBuffer(&buffer); 6432 } 6433 } else { 6434 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6435 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6436 6437 // reallocate buffer if needed 6438 if (mBufFrameSize != 0 && mBufFrames < frames) { 6439 free(mBuf); 6440 mBufFrames = frames; 6441 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6442 } 6443 // resampler accumulates, but we only have one source track 6444 memset(mBuf, 0, frames * mBufFrameSize); 6445 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6446 // format convert to destination buffer 6447 convertResampler(dst, mBuf, frames); 6448 } 6449 return frames; 6450} 6451 6452status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6453 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6454 uint32_t srcSampleRate, 6455 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6456 uint32_t dstSampleRate) 6457{ 6458 // quick evaluation if there is any change. 6459 if (mSrcFormat == srcFormat 6460 && mSrcChannelMask == srcChannelMask 6461 && mSrcSampleRate == srcSampleRate 6462 && mDstFormat == dstFormat 6463 && mDstChannelMask == dstChannelMask 6464 && mDstSampleRate == dstSampleRate) { 6465 return NO_ERROR; 6466 } 6467 6468 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6469 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6470 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6471 const bool valid = 6472 audio_is_input_channel(srcChannelMask) 6473 && audio_is_input_channel(dstChannelMask) 6474 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6475 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6476 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6477 ; // no upsampling checks for now 6478 if (!valid) { 6479 return BAD_VALUE; 6480 } 6481 6482 mSrcFormat = srcFormat; 6483 mSrcChannelMask = srcChannelMask; 6484 mSrcSampleRate = srcSampleRate; 6485 mDstFormat = dstFormat; 6486 mDstChannelMask = dstChannelMask; 6487 mDstSampleRate = dstSampleRate; 6488 6489 // compute derived parameters 6490 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6491 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6492 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6493 6494 // do we need to resample? 6495 delete mResampler; 6496 mResampler = NULL; 6497 if (mSrcSampleRate != mDstSampleRate) { 6498 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6499 mSrcChannelCount, mDstSampleRate); 6500 mResampler->setSampleRate(mSrcSampleRate); 6501 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6502 } 6503 6504 // are we running legacy channel conversion modes? 6505 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6506 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6507 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6508 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6509 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6510 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6511 6512 // do we need to process in float? 6513 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6514 6515 // do we need a staging buffer to convert for destination (we can still optimize this)? 6516 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6517 if (mResampler != NULL) { 6518 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6519 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6520 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6521 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6522 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6523 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6524 } else { 6525 mBufFrameSize = 0; 6526 } 6527 mBufFrames = 0; // force the buffer to be resized. 6528 6529 // do we need an input converter buffer provider to give us float? 6530 delete mInputConverterProvider; 6531 mInputConverterProvider = NULL; 6532 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6533 mInputConverterProvider = new ReformatBufferProvider( 6534 audio_channel_count_from_in_mask(mSrcChannelMask), 6535 mSrcFormat, 6536 AUDIO_FORMAT_PCM_FLOAT, 6537 256 /* provider buffer frame count */); 6538 } 6539 6540 // do we need a remixer to do channel mask conversion 6541 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6542 (void) memcpy_by_index_array_initialization_from_channel_mask( 6543 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6544 } 6545 return NO_ERROR; 6546} 6547 6548void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6549 void *dst, const void *src, size_t frames) 6550{ 6551 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6552 if (mBufFrameSize != 0 && mBufFrames < frames) { 6553 free(mBuf); 6554 mBufFrames = frames; 6555 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6556 } 6557 // do we need to do legacy upmix and downmix? 6558 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6559 void *dstBuf = mBuf != NULL ? mBuf : dst; 6560 if (mIsLegacyUpmix) { 6561 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6562 (const float *)src, frames); 6563 } else /*mIsLegacyDownmix */ { 6564 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6565 (const float *)src, frames); 6566 } 6567 if (mBuf != NULL) { 6568 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6569 frames * mDstChannelCount); 6570 } 6571 return; 6572 } 6573 // do we need to do channel mask conversion? 6574 if (mSrcChannelMask != mDstChannelMask) { 6575 void *dstBuf = mBuf != NULL ? mBuf : dst; 6576 memcpy_by_index_array(dstBuf, mDstChannelCount, 6577 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6578 if (dstBuf == dst) { 6579 return; // format is the same 6580 } 6581 } 6582 // convert to destination buffer 6583 const void *convertBuf = mBuf != NULL ? mBuf : src; 6584 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6585 frames * mDstChannelCount); 6586} 6587 6588void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6589 void *dst, /*not-a-const*/ void *src, size_t frames) 6590{ 6591 // src buffer format is ALWAYS float when entering this routine 6592 if (mIsLegacyUpmix) { 6593 ; // mono to stereo already handled by resampler 6594 } else if (mIsLegacyDownmix 6595 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6596 // the resampler outputs stereo for mono input channel (a feature?) 6597 // must convert to mono 6598 downmix_to_mono_float_from_stereo_float((float *)src, 6599 (const float *)src, frames); 6600 } else if (mSrcChannelMask != mDstChannelMask) { 6601 // convert to mono channel again for channel mask conversion (could be skipped 6602 // with further optimization). 6603 if (mSrcChannelCount == 1) { 6604 downmix_to_mono_float_from_stereo_float((float *)src, 6605 (const float *)src, frames); 6606 } 6607 // convert to destination format (in place, OK as float is larger than other types) 6608 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6609 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6610 frames * mSrcChannelCount); 6611 } 6612 // channel convert and save to dst 6613 memcpy_by_index_array(dst, mDstChannelCount, 6614 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6615 return; 6616 } 6617 // convert to destination format and save to dst 6618 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6619 frames * mDstChannelCount); 6620} 6621 6622bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6623 status_t& status) 6624{ 6625 bool reconfig = false; 6626 6627 status = NO_ERROR; 6628 6629 audio_format_t reqFormat = mFormat; 6630 uint32_t samplingRate = mSampleRate; 6631 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6632 // possible that we are > 2 channels, use channel index mask 6633 if (channelMask == AUDIO_CHANNEL_INVALID && mChannelCount <= FCC_8) { 6634 audio_channel_mask_for_index_assignment_from_count(mChannelCount); 6635 } 6636 6637 AudioParameter param = AudioParameter(keyValuePair); 6638 int value; 6639 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6640 // channel count change can be requested. Do we mandate the first client defines the 6641 // HAL sampling rate and channel count or do we allow changes on the fly? 6642 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6643 samplingRate = value; 6644 reconfig = true; 6645 } 6646 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6647 if (!audio_is_linear_pcm((audio_format_t) value)) { 6648 status = BAD_VALUE; 6649 } else { 6650 reqFormat = (audio_format_t) value; 6651 reconfig = true; 6652 } 6653 } 6654 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6655 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6656 if (!audio_is_input_channel(mask) || 6657 audio_channel_count_from_in_mask(mask) > FCC_8) { 6658 status = BAD_VALUE; 6659 } else { 6660 channelMask = mask; 6661 reconfig = true; 6662 } 6663 } 6664 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6665 // do not accept frame count changes if tracks are open as the track buffer 6666 // size depends on frame count and correct behavior would not be guaranteed 6667 // if frame count is changed after track creation 6668 if (mActiveTracks.size() > 0) { 6669 status = INVALID_OPERATION; 6670 } else { 6671 reconfig = true; 6672 } 6673 } 6674 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6675 // forward device change to effects that have requested to be 6676 // aware of attached audio device. 6677 for (size_t i = 0; i < mEffectChains.size(); i++) { 6678 mEffectChains[i]->setDevice_l(value); 6679 } 6680 6681 // store input device and output device but do not forward output device to audio HAL. 6682 // Note that status is ignored by the caller for output device 6683 // (see AudioFlinger::setParameters() 6684 if (audio_is_output_devices(value)) { 6685 mOutDevice = value; 6686 status = BAD_VALUE; 6687 } else { 6688 mInDevice = value; 6689 // disable AEC and NS if the device is a BT SCO headset supporting those 6690 // pre processings 6691 if (mTracks.size() > 0) { 6692 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6693 mAudioFlinger->btNrecIsOff(); 6694 for (size_t i = 0; i < mTracks.size(); i++) { 6695 sp<RecordTrack> track = mTracks[i]; 6696 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6697 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6698 } 6699 } 6700 } 6701 } 6702 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6703 mAudioSource != (audio_source_t)value) { 6704 // forward device change to effects that have requested to be 6705 // aware of attached audio device. 6706 for (size_t i = 0; i < mEffectChains.size(); i++) { 6707 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6708 } 6709 mAudioSource = (audio_source_t)value; 6710 } 6711 6712 if (status == NO_ERROR) { 6713 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6714 keyValuePair.string()); 6715 if (status == INVALID_OPERATION) { 6716 inputStandBy(); 6717 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6718 keyValuePair.string()); 6719 } 6720 if (reconfig) { 6721 if (status == BAD_VALUE && 6722 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6723 audio_is_linear_pcm(reqFormat) && 6724 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6725 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6726 audio_channel_count_from_in_mask( 6727 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6728 status = NO_ERROR; 6729 } 6730 if (status == NO_ERROR) { 6731 readInputParameters_l(); 6732 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6733 } 6734 } 6735 } 6736 6737 return reconfig; 6738} 6739 6740String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6741{ 6742 Mutex::Autolock _l(mLock); 6743 if (initCheck() != NO_ERROR) { 6744 return String8(); 6745 } 6746 6747 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6748 const String8 out_s8(s); 6749 free(s); 6750 return out_s8; 6751} 6752 6753void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event) { 6754 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6755 6756 desc->mIoHandle = mId; 6757 6758 switch (event) { 6759 case AUDIO_INPUT_OPENED: 6760 case AUDIO_INPUT_CONFIG_CHANGED: 6761 desc->mPatch = mPatch; 6762 desc->mChannelMask = mChannelMask; 6763 desc->mSamplingRate = mSampleRate; 6764 desc->mFormat = mFormat; 6765 desc->mFrameCount = mFrameCount; 6766 desc->mLatency = 0; 6767 break; 6768 6769 case AUDIO_INPUT_CLOSED: 6770 default: 6771 break; 6772 } 6773 mAudioFlinger->ioConfigChanged(event, desc); 6774} 6775 6776void AudioFlinger::RecordThread::readInputParameters_l() 6777{ 6778 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6779 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6780 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6781 if (mChannelCount > FCC_8) { 6782 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6783 } 6784 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6785 mFormat = mHALFormat; 6786 if (!audio_is_linear_pcm(mFormat)) { 6787 ALOGE("HAL format %#x is not linear pcm", mFormat); 6788 } 6789 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6790 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6791 mFrameCount = mBufferSize / mFrameSize; 6792 // This is the formula for calculating the temporary buffer size. 6793 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6794 // 1 full output buffer, regardless of the alignment of the available input. 6795 // The value is somewhat arbitrary, and could probably be even larger. 6796 // A larger value should allow more old data to be read after a track calls start(), 6797 // without increasing latency. 6798 // 6799 // Note this is independent of the maximum downsampling ratio permitted for capture. 6800 mRsmpInFrames = mFrameCount * 7; 6801 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6802 free(mRsmpInBuffer); 6803 6804 // TODO optimize audio capture buffer sizes ... 6805 // Here we calculate the size of the sliding buffer used as a source 6806 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6807 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6808 // be better to have it derived from the pipe depth in the long term. 6809 // The current value is higher than necessary. However it should not add to latency. 6810 6811 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6812 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6813 6814 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6815 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6816} 6817 6818uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6819{ 6820 Mutex::Autolock _l(mLock); 6821 if (initCheck() != NO_ERROR) { 6822 return 0; 6823 } 6824 6825 return mInput->stream->get_input_frames_lost(mInput->stream); 6826} 6827 6828uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6829{ 6830 Mutex::Autolock _l(mLock); 6831 uint32_t result = 0; 6832 if (getEffectChain_l(sessionId) != 0) { 6833 result = EFFECT_SESSION; 6834 } 6835 6836 for (size_t i = 0; i < mTracks.size(); ++i) { 6837 if (sessionId == mTracks[i]->sessionId()) { 6838 result |= TRACK_SESSION; 6839 break; 6840 } 6841 } 6842 6843 return result; 6844} 6845 6846KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6847{ 6848 KeyedVector<int, bool> ids; 6849 Mutex::Autolock _l(mLock); 6850 for (size_t j = 0; j < mTracks.size(); ++j) { 6851 sp<RecordThread::RecordTrack> track = mTracks[j]; 6852 int sessionId = track->sessionId(); 6853 if (ids.indexOfKey(sessionId) < 0) { 6854 ids.add(sessionId, true); 6855 } 6856 } 6857 return ids; 6858} 6859 6860AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6861{ 6862 Mutex::Autolock _l(mLock); 6863 AudioStreamIn *input = mInput; 6864 mInput = NULL; 6865 return input; 6866} 6867 6868// this method must always be called either with ThreadBase mLock held or inside the thread loop 6869audio_stream_t* AudioFlinger::RecordThread::stream() const 6870{ 6871 if (mInput == NULL) { 6872 return NULL; 6873 } 6874 return &mInput->stream->common; 6875} 6876 6877status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6878{ 6879 // only one chain per input thread 6880 if (mEffectChains.size() != 0) { 6881 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6882 return INVALID_OPERATION; 6883 } 6884 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6885 chain->setThread(this); 6886 chain->setInBuffer(NULL); 6887 chain->setOutBuffer(NULL); 6888 6889 checkSuspendOnAddEffectChain_l(chain); 6890 6891 // make sure enabled pre processing effects state is communicated to the HAL as we 6892 // just moved them to a new input stream. 6893 chain->syncHalEffectsState(); 6894 6895 mEffectChains.add(chain); 6896 6897 return NO_ERROR; 6898} 6899 6900size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6901{ 6902 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6903 ALOGW_IF(mEffectChains.size() != 1, 6904 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6905 chain.get(), mEffectChains.size(), this); 6906 if (mEffectChains.size() == 1) { 6907 mEffectChains.removeAt(0); 6908 } 6909 return 0; 6910} 6911 6912status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6913 audio_patch_handle_t *handle) 6914{ 6915 status_t status = NO_ERROR; 6916 6917 // store new device and send to effects 6918 mInDevice = patch->sources[0].ext.device.type; 6919 mPatch = *patch; 6920 for (size_t i = 0; i < mEffectChains.size(); i++) { 6921 mEffectChains[i]->setDevice_l(mInDevice); 6922 } 6923 6924 // disable AEC and NS if the device is a BT SCO headset supporting those 6925 // pre processings 6926 if (mTracks.size() > 0) { 6927 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6928 mAudioFlinger->btNrecIsOff(); 6929 for (size_t i = 0; i < mTracks.size(); i++) { 6930 sp<RecordTrack> track = mTracks[i]; 6931 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6932 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6933 } 6934 } 6935 6936 // store new source and send to effects 6937 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6938 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6939 for (size_t i = 0; i < mEffectChains.size(); i++) { 6940 mEffectChains[i]->setAudioSource_l(mAudioSource); 6941 } 6942 } 6943 6944 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6945 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6946 status = hwDevice->create_audio_patch(hwDevice, 6947 patch->num_sources, 6948 patch->sources, 6949 patch->num_sinks, 6950 patch->sinks, 6951 handle); 6952 } else { 6953 char *address; 6954 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 6955 address = audio_device_address_to_parameter( 6956 patch->sources[0].ext.device.type, 6957 patch->sources[0].ext.device.address); 6958 } else { 6959 address = (char *)calloc(1, 1); 6960 } 6961 AudioParameter param = AudioParameter(String8(address)); 6962 free(address); 6963 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 6964 (int)patch->sources[0].ext.device.type); 6965 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 6966 (int)patch->sinks[0].ext.mix.usecase.source); 6967 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6968 param.toString().string()); 6969 *handle = AUDIO_PATCH_HANDLE_NONE; 6970 } 6971 6972 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6973 6974 return status; 6975} 6976 6977status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6978{ 6979 status_t status = NO_ERROR; 6980 6981 mInDevice = AUDIO_DEVICE_NONE; 6982 6983 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6984 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6985 status = hwDevice->release_audio_patch(hwDevice, handle); 6986 } else { 6987 AudioParameter param; 6988 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 6989 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6990 param.toString().string()); 6991 } 6992 return status; 6993} 6994 6995void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6996{ 6997 Mutex::Autolock _l(mLock); 6998 mTracks.add(record); 6999} 7000 7001void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7002{ 7003 Mutex::Autolock _l(mLock); 7004 destroyTrack_l(record); 7005} 7006 7007void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7008{ 7009 ThreadBase::getAudioPortConfig(config); 7010 config->role = AUDIO_PORT_ROLE_SINK; 7011 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7012 config->ext.mix.usecase.source = mAudioSource; 7013} 7014 7015} // namespace android 7016